U.S. patent number 6,728,345 [Application Number 09/876,979] was granted by the patent office on 2004-04-27 for system and method for recording and storing telephone call information.
This patent grant is currently assigned to Dictaphone Corporation. Invention is credited to David A. Glowny, Phil Min Ni, John E. Richter.
United States Patent |
6,728,345 |
Glowny , et al. |
April 27, 2004 |
**Please see images for:
( Certificate of Correction ) ** |
System and method for recording and storing telephone call
information
Abstract
A system and method for monitoring a telephone switching
environment. In a preferred embodiment the system and method
identify telephone call segments that relate to one telephone call
and construct a data representation of a lifetime of the telephone
call, using data regarding telephony events associated with the
telephone call segments of the telephone call. The system and
method are also capable of using the data representation to display
a graphical representation of a lifetime of a telephone call.
Inventors: |
Glowny; David A. (Milford,
CT), Ni; Phil Min (Danbury, CT), Richter; John E.
(Trumbull, CT) |
Assignee: |
Dictaphone Corporation
(Stratford, CT)
|
Family
ID: |
23280393 |
Appl.
No.: |
09/876,979 |
Filed: |
June 8, 2001 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
Issue Date |
|
|
328299 |
Jun 8, 1999 |
6249570 |
|
|
|
Current U.S.
Class: |
379/88.22;
G9B/31; 379/111; 379/202.01 |
Current CPC
Class: |
G11B
31/00 (20130101); H04M 3/36 (20130101); H04M
3/42221 (20130101); H04M 3/2218 (20130101); H04M
3/2281 (20130101); H04M 2203/301 (20130101); H04M
3/42314 (20130101); H04M 3/42323 (20130101); H04M
3/51 (20130101); H04M 3/5175 (20130101) |
Current International
Class: |
G11B
31/00 (20060101); H04M 3/36 (20060101); H04M
3/42 (20060101); H04M 3/22 (20060101); H04M
3/50 (20060101); H04M 3/51 (20060101); H04M
001/04 () |
Field of
Search: |
;379/67.1,68,88.09,88.11,88.22,88.25,93.12,93.17,93.23,111,112.01,112.06,116,202.01,265.03,265.07,266.1,267,309
;360/5,6,55 |
References Cited
[Referenced By]
U.S. Patent Documents
Primary Examiner: Weaver; Scott L.
Attorney, Agent or Firm: Howrey Simon Arnold & White,
LLP Meola; Anthony L.
Parent Case Text
This is a continuation of application Ser. No. 09/328,299, filed
Jun. 8, 1999 now U.S. Pat. No. 6,249,570.
Claims
What is claimed is:
1. A system for recording information regarding telephone calls
with three or more participants and comprising one or more
telephone call segments, comprising: (a) a first memory having one
or more locations storing audio data of telephone call segments (b)
a second memory having one or more locations storing data regarding
telephony events associated with the telephone call segments; and
(c) a processor programmed to: (i) identify telephone call segments
that relate to the same telephone call and (ii) construct a data
representation of lifetimes of the telephone calls that have three
or more participants, wherein said data representations are
constructed using data regarding telephony events associated with
the telephone call segments.
2. The system of claim 1 wherein the data representation of each
telephone call comprises (i) a list of participants in the
telephone call; (ii) a list of telephony events regarding the call;
(iii) a list containing the time each telephony event occurred; and
(iv) the start and end time of the call.
3. The system of claim 1 wherein the data representation of each
telephone call comprises, for each segment of the call, the
location of the stored audio data of that segment.
4. The system of claim 1 wherein the first memory and the second
memory are the same.
5. The system of claim 1 wherein the processor is comprised of a
plurality of physically separated components.
6. The system of claim 3 wherein the location of the stored audio
data of each segment comprises a location of a .WAV file containing
the audio data.
7. The system of claim 6 wherein the data representation of a
telephone call further comprises an offset within the .WAV file to
the start of the stored audio data.
8. The system of claim 1 wherein the data regarding telephony
events is received from a plurality of sources connected to a
telephone switching environment.
9. The system of claim 1 further comprising display software that
uses said data representation to display a graphical representation
of said telephone call.
10. The system of claim 2 further comprising display software that
uses a data representation of a telephone call to display a
graphical representation of said telephone call.
11. The system of claim 10 wherein the graphical representation
comprises a representation of each segment of the call.
12. The system of claim 10 wherein the graphical representation
comprises a representation of the length of time of each segment of
the call.
13. The system of claim 9 wherein the display software further
displays a table comprising data from the data representation.
14. A method for recording information regarding telephone calls
with three or more participants and comprising one or more
participants and comprising one or more telephone call segments,
comprising: (a) receiving audio data regarding one or more
telephone call segments; (b) receiving data regarding telephony
events associated with said telephone call segments; (c) storing
the received audio data regarding telephone call segments; (d)
storing the received data regarding telephone events associated
with said telephone call segments; (e) identifying telephone call
segments that relate to the same telephone call; and (f)
constructing data representations of lifetimes of telephone calls,
wherein said data representations are constructed using data
regarding telephony events associated with telephone call
segments.
15. The method of claim 14 wherein each data representation of a
telphone call comprises: (i) a list of participants in the
telephone call; (ii) a list of telephony events regarding the call;
(iii) a list containing the time each telephony event occurred; and
(iv) the start and end time of the call.
16. The method of claim 14 wherein each data representation of a
telephone call comprises, for each segment of the call, a location
of stored audio data if that segment.
17. The method of claim 14 wherein the received audio data and the
data regarding telephony events are stored in the same memory.
18. The method of claim 14 wherein each data representation is
constructed by a plurality of physically separated processors.
19. The method of claim 16 wherein the location of the stored audio
data of each segment comprises a location of a .WAV file containng
the audio data.
20. The method of claim 19 wherein a data representation further
comprises an offset within the .WAV file to start of the stored
audio data.
21. The method of claim 14 wherein data regarding telephony events
is received from a plurality of sources connected to a telephone
switching environment.
22. The method of claim 14 further comprising the step of using a
data representation of a telephone call to display a graphical
representation of the telephone call.
23. The method of claim 15 further comprising the step of using
said data representationn of a telephone call to display a
graphical representation of the telephone call.
24. The method of claim 23 wherein the graphical representation
comprises a representation of each segment of the call.
25. The method of claim 23 wherein the graphical representation
comprises a representation of the length of time of each segment of
the call.
26. The method of claim 22 further comprising the step of
displaying a table comprising data from the data
representation.
27. A system for recording information regarding telephone calls
comprising one or more telephone call segments, wherein said calls
comprise calls wherein at least one participant participates in a
plurality of segments, comprising: (a) a first memory having one or
more locations storing audio data regarding telephone call
segments; (b) a second memory having one or more locations storing
data regarding telephony events associated with telephone call
segments; and (c) a processor programmed to: (i) identify telephone
call segments that relate to the same telephone call; (ii) identify
multiple call segments that have the same participant; and (iii)
construct data representations of lifetimes of telephone calls
using data regarding telephony events associated with telephone
call segments.
28. The system of claim 27 wherein a data representation of a
telephone call comprises: (i) a list of participants in the
telephone call; (ii) a list of telephony events regarding the call;
(iii) a list containing the time each telephony event occurred; and
(iv) the start and end time of the call.
29. The system of claim 27 wherein each data representation of a
telephone call comprises, for each segment of the call, a location
of the stored audio data of that segment.
30. The system of claim 27 wherein the first memory and the second
memory are the same.
31. The system of claim 27 wherein the processor is comprised of a
plurality of physically separated components.
32. The system of claim 29 wherein the location of the stored audio
data of each segment comprises a location of a .WAV file containing
the audio data.
33. The system of claim 32 wherein a data representation of a
telephone call further comprises an offset within the .WAV file to
the start of the stored audio data.
34. The system of claim 27 wherein data regarding telephony events
is received from a plurality of sources connected to a telephone
switching environment.
35. The system of claim 27 further comprising display software that
uses a data representation of a telephone call to display a
graphical representation of said telephone call.
36. The system of claim 28 further comprising display software that
uses a data representation of a telephone call to display a
graphical representation of said telephone call.
37. The system of claim 36 wherein the graphical representation
comprises a representation of each segment of the call.
38. The system of claim 36 wherein the graphical representation
comprises a representation of the length of time of each segment of
the call.
39. The system of claim 35 wherein the display software further
displays a table comprising data from the data representation.
40. A method for recording information regarding telephone calls
comprising one or more telephone call segments, wherein said calls
comprise calls wherein at least one participant paprticipates in a
plurality of segments, comprising: (a) receiving audio data
regarding one or more telephone call segments and data regarding
telephone events associated with said telephone call segments; (b)
storing the received audio data regarding telephone call segments;
(c) storing the received data regarding telephony events associated
with said telephone call segments; (d) identifying telephone call
segments that relate to one telephone call; (e) identifying
multiple call segments that have the same participant; and (f)
constructing data representations of lifetimes of telephone calls,
wherein each data representation of a telephone call is constructed
using data regarding telephony events associated with the telephone
call segments of the telephone call.
41. The method of claim 40 wherein a data representation of a
telephone call comprises: (i) a list of participants in the
telephone call; (ii) a list of telephony events regarding the call;
(iii) a list containing the time each telephony event occurred; and
(iv) the start and end time of the call.
42. The method of claim 40 wherein a data representation of a
telephone call comprises, for each segment of the call, the
location of the stored audio data of that segment.
43. The method of claim 40 wherein the received audio data and the
data regarding telephony events is stored in the same memory.
44. The method of claim 40 wherein a data representation of a
telephone call is constructed by a plurality of physically
separated processors.
45. The method of claim 42 wherein a location of stored audio data
of each segment comprises the location of a .WAV file containing
the audio data.
46. The method of claim 45 wherein a data representation of a
telephone call further comprises an offset within the .WAV file to
the start of the stored audio data.
47. The method of claim 40 wherein data regarding telephony events
is received from a plurality of sources connected to a telephone
switching environment.
48. The method of claim 40 further comprising the step of using a
data representation of a telephone call to display a graphical
representation of said telephone call.
49. The method of claim 41 further comprising the step of using a
data representation of a telephone call to display a graphical
representation of said telephone call.
50. The method of claim 49 wherein the graphical representation
comprises a representation of each segment of the call.
51. The method of claim 49 wherein the graphical representation
comprises a representation of the length of time of each segment of
the call.
52. The method of claim 48 further comprising the step of
displaying a table comprising data from the data representation.
Description
FIELD OF THE INVENTION
This invention relates generally to computer-aided data recording.
In particular, it relates to computer-aided monitoring and
recording of telephone calls.
BACKGROUND OF THE INVENTION
Telephone call monitoring systems are used in a variety of
contexts, including emergency dispatch centers and commercial call
centers. In many currently available call monitoring systems, a
multitude of audio input sources ("channels") are monitored and
recordedby a single hardware unit, and the audio recordings are
saved and organized according to the input channel, date, time and
duration. The capacity of the recording unit can be expanded to
handle a larger number of channels by combining several recording
units into a system using a local area network (LAN). Because
retrieval is only possible using basic search criteria (recording
unit, channel, date, time and duration), it is often difficult to
locate a particular audio recording that is of interest. When there
is a need to search for a recording according to search criteria
that are not directly supported by simple voice recording, locating
a specific recording may require tedious and repetitive searching.
For example, if there is a need to find a specific customer's call
to resolve a disputed transaction, the recording unit or channel
that carried the original call might not be known, so the searcher
would be forced to manually play back many calls before finding the
correct one.
With the advent of computer telephony integration (CTI), it is now
possible to monitor a data link that supplies more information
about telephone calls, in addition to simple voice recording. In a
typical CTI system a telephone switch or private branch exchange
(PBX) provides an interface suitable for processing by a computer,
and expanded information about telephone calls is made available
through this interface as the calls occur. Data fields that are
available within this expanded information may include the external
telephone number of the calling party, as well as identification
numbers to help associate a series of events pertaining to the same
call. With such a data link being used alongside a voice recording
system, the search and retrieval system can be supplemented by
constructing a database that combines the previously discussed
basic search criteria with enhanced search criteria (based upon
information obtained through a CTI data link) such as: telephone
numbers of parties involved in the call; Caller ID (CLID) or
Automatic Number Identification (ANI); Dialed Number Identification
Service (DNIS); or the Agent ID Number of the Customer Service
Representative.
As shown in FIG. 2, with suitable equipment for tapping into a
voice communications line, a recording unit can intercept telephone
call traffic using two methods. By attaching wires for recording
channels on each extension within a call center, the traffic can be
intercepted and recorded as it passes between the PBX and the agent
telephone set. This first method is known as "station-side"
recording 180. Alternatively, by attaching equipment on the trunk
lines between the PBX and Public Switched Telephone Network (PSTN),
the traffic can be intercepted at its point of entry into the call
center before the calls are dispatched by the PBX. This second
method is known as "trunk-side" recording 170. Since businesses
usually have more agent telephone sets than trunk lines, a
"trunk-side" solution is likely to require less recording equipment
and thus be less expensive. Another significant point for
consideration is that "trunk-side" provides access only to external
inbound or outbound calls, which are those typically involving
customers of a business, whereas "station-side" also provides
access to internal calls between agents (which may or may not
relate to an external customer's transaction).
With respect to data links to provide call information to
computers, there are typically two different categories of links
from the PBX available. Some older links use interfaces such as
SMDR (Station Message Detail Recording) or CDR (Call Detail
Recording) that provide summary information about telephone calls
in a line-oriented text format. Both acronyms refer to essentially
the same type of system. Information from these links is generally
provided after the call has concluded, and as such is suitable for
billing applications or traffic analysis software. Many newer links
use real-time interfaces that are designed to supply a series of
events while a telephone call is still active within the PBX, to
enable computer and multimedia systems to respond and interact with
an external caller. The information provided by such real-time
links is typically much more detailed than that provided by
SMDR.
The detailed information and real-time nature of a CTI link is
particularly important when building a recording system that is
intended to react to telephone calls as they occur and to
dynamically select which calls ought to be recorded or discarded.
CTI-supplied information is also important when building a
recording system that is intended to capture the full history of a
telephone call, including recording the different agents who were
involved in the conversation and how the call was held, transferred
or conferenced. Likewise, real-time information is important in a
system that intends to support (a) a live display of active calls,
and (b) the capability for a user to listen and monitor the live
audio traffic.
A "trunk-side" solution based upon voice recording alone will not
satisfy the above requirements in a practical manner, since
telephone calls are assigned to trunks dynamically as needed to
handle the traffic. What trunk channel a particular call will be
carried on cannot be predicted in advance. Without information to
associate a logical telephone call with a physical recording of
audio from a trunk channel, a user might have to search and
retrieve many recordings before finding the one that is of
interest. Moreover, in a system designed to make use of the
enhanced search criteria provided by a data link, it would not be
possible to programmatically associate the search data with the
voice recording without information about the trunk channel where
the call occurred.
This problem can be avoided as long as the data link provides
sufficient information about the trunk channels being used for each
call. Unfortunately, some PBX environments do not supply this
critical information about trunk channels within the data provided
on the real-time CTI link. For example, this problem is manifested
by the Lucent Technologies DEFINITY G3 PBX, which is a commonly
used telephone switch in North America. While the Lucent G3 PBX
provides trunk channel information through its SMDR link, that
information is not available until after the conclusion of the
call. This presents a problem for system features and capabilities
dependent upon real-time data. The real-time data link provided by
the Lucent G3 PBX does not provide the necessary information about
trunk channels. There is thus a need for a system which is capable
of simultaneously monitoring both the SMDR link and the real-time
CTI link, gathering information about calls from both sources, and
combining that information into a single data model of the
telephony activity within the call center. There is a further need
for a system that combines the data model with information
concerning the location of call recordings, resulting in a "master
call record" that contains data matching each call with the
segments of which it is comprised, and matching the data for each
segment with the location of the recording of that segment. Such a
system would facilitate monitoring, recording, and playing back
complete telephone calls.
SUMMARY OF THE INVENTION
The present invention is directed to a system and method that is
capable of simultaneously monitoring two or more data links,
gathering information about calls from those data links, combining
that information into a single data model of the telephony activity
within the call center, and combining the data model with
information concerning the location of call recordings, resulting
in a "master call record" that contains data matching each call
with the segments of which it is comprised, and matching the data
for each segment with the location of the recording of that
segment.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a block diagram of the system of this invention in a
preferred embodiment.
FIG. 2 illustrates the difference between trunk-side and
station-side recording.
FIG. 3 shows a line-chart that illustrates various parties involved
in a complex call.
FIG. 4 shows a schematic block diagram of a preferred embodiment
for translating, summarizing and normalizing signals received from
both an SMDR link and a Dialogic CT-Connect CTI service.
FIG. 5 illustrates the steps by which the translation module
CtiCtc.exe integrates the data received from the CTI and SMDR
links.
FIG. 6 illustrates how the CTI Server can be viewed as a set of
logically distinct layers that deal with translating and
distributing CTI events.
FIG. 7 illustrates how, in addition to telephony events, the CTI
Server 710 is responsible for supplying certain metadata regarding
agent events to the System Controller 130.
FIG. 8 shows the layout of the CTI Server.
FIG. 9 shows a version of CtiCtc.exe configured to work with a
Lucent Telephony Services interface (and thus called CtiLts.exe
instead of CtiCtc.exe).
FIG. 10 depicts key elements of the data model used in a-preferred
embodiment.
FIG. 11 illustrates three distinct layers of the CTI Server in a
preferred embodiment.
FIG. 12 shows in block diagram form several threads of the CTI
Server in a preferred embodiment that implement three distinct
layers of processing (data collection, data normalization, and
message emission).
FIG. 13 illustrates the program logic flow of the analyzer layer of
the preferred embodiment.
FIG. 14 depicts the flow of information within the recording system
of this invention in a preferred embodiment.
FIG. 15 shows how a recording unit operating with only voice
signaling to guide the creation of its call records could make a
number of fragmented audio segments.
FIG. 16 shows a graphical user interface used in the preferred
embodiment.
FIG. 16A shows a system containing a CTI Server and a Recorder in a
specific embodiment of the present invention.
FIG. 16B is a table illustrating descriptive information from the
CTI Server used in a specific embodiment.
FIG. 17 illustrates steps in the creation of a Master Call Record
used in a specific embodiment.
FIG. 18 shows the processing threads and data structures that
comprise the CRG module in accordance with the present
invention.
FIG. 19 illustrates the class diagram of the Call Record Generator
used in a specific embodiment.
FIGS. 20, 20A, 20B, 21, 22, 22A, 22B, and 22C illustrate the
operation of the Stream Control Manager.
DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS
The present invention is directed to a communication recording
system and method. Generally, the functionality of the system
involves tapping into activity on a PBX (Private Branch Exchange)
by intercepting audio on either the trunk or station side of a
telephone call. The tapped audio is then redirected as input to a
channel on a Digital Signal Processor (DSP) based voice processing
board, which in turn is digitized into program addressable buffers.
The recorded digitized audio is then combined with descriptive
information ("metadata") obtained through a Computer Telephony
Integration (CTI) communications link with the PBX, and stored as a
single manageable unit ("Voicedata") to facilitate its subsequent
search and retrieval. The system uses modular architecture in both
its hardware and software, so that any one component can be
replaced or upgraded without affecting the rest of the system.
In a preferred embodiment the communications recording system
comprises multiple rack-mountable computer-processing servers (such
as the Compaq ProLiant 1600 R), using a multi-tasking operating
system (e.g., Microsoft Windows NT), DSP voice processing boards
(e.g., Dialogic D/160SC), and a distributed set of software
components available from Dictaphone Corporation. In a specific
embodiment directed to the smallest configuration, all of these
components may reside in a single computer-processing server. In
other preferred embodiments, related components are typically
packaged in combinations and the entire system spans multiple
servers that coordinate processing through a Local Area Network
(LAN).
In this preferred configuration, the overall system generally
comprises CTI Servers, Voice Servers, a Central Database Server,
and User Workstations. CTI servers generally use a set of
components to manage a data communications link with a telephone
switch environment, to obtain notification of calls as they occur,
along with the descriptive information about the calls (e.g.,
source and destination telephone numbers). The Voice Servers use a
set of components to collect audio recordings, manage their
storage, and facilitate their playback through the LAN. The Central
Database Server uses a set of components to manage system-wide
search and retrieval of recorded calls. User Workstations are
typically desktop computers that use a set of components to allow a
person to submit requests to search and retrieve recorded calls for
playback and to control automatically scheduled functions within
the recording system.
FIG. 1 shows in a block diagram from components of the system of
this invention in a preferred embodiment. Data enters the recording
system from a variety of sources. These sources can include a PBX
100, CTI middleware 105, ISDN lines 110, or other input sources
115. It will thus be appreciated that the system of the present
invention can be used for monitoring and recording of information
about any type of electronic communications. For simplicity, the
following discussion uses the term Telephone calls. However, it is
intended that term covers any electronic communication unless
specified otherwise expressly.
Data from data sources 100, 105, 110 or 115 is transmitted to one
or more CTI Translation Modules 165, which translates input data
into a common format. The data is then sent to a CTI Message Router
120, which distributes the data onward to appropriate components of
the system.
Audio Recorders 145 may be used for passive trunk-side 170 and
extension-side 180 recording on a pre-determined static set of
devices, as well as dynamically initiated recording of specific
devices according to scheduling criteria through the Service
Observance feature 185 provided by a telephone switch environment.
The recordings are stored on an audio storage device 140. A Call
Record Generator 150 matches data from the Audio Recorders 145 with
data sent by the CTI Message Router 120 to create a Master Call
Record (MCR) for each telephone call. The MCRs are stored in a
Voicedata Storage module 155. One or more User Workstations 160 use
the MCRs to reconstruct and play back complete or partial phone
conversations stored in the audio storage device 140. A Scheduling
and Control Services module 130 controls the Audio Recorders 145
and communicates with User Workstation 160. The Scheduling and
Control Services module is responsible for starting and stopping
the audio recording activity, according to pre-defined rules that
are dependent upon time data provided by the Time Service 115 and
CTI information. As the system components are packaged in the
typical configuration, the CTI translation modules 165 and CTI
message router 120 are co-resident upon a computer-processing
server called the CTI Server 710. In a similar fashion, the
combined set of components including the Time Service 125,
Scheduling & Control Services 130, Audio Recorder 145, Audio
Storage 140, and Call Record Generator 150, in a specific
embodiment can be co-resident upon a computer-processing server
called the Voice Server 124. The Voicedata storage 155 resides
within a computer-processing server called the Central Database
Server. The specialized application software for the User
Workstation 160 resides upon desktop computers that use, in a
preferred embodiment, Microsoft Windows 95, Windows 98 or Windows
NT operating systems.
As noted above, in a specific embodiment the CTI Server comprises
two main modules: a CTI translation module (such as the software
program CtiCtc.exe, CtiLts.exe, and other translation modules) and
a CTI Message Router module (such as the software program
CtiServ.exe discussed below, or its equivalent). In a specific
embodiment, the CTI Server may have several translation modules,
for example, one for each PBX interface, or for each vendor API
layer. As shown in FIG. 1, the CTI Server of the preferred
embodiment accepts data from a PBX or similar equipment in a
telephone switch environment, and can use both real-time CTI
communications links and asynchronous information sources such the
Station Message Detail Recording (SMDR) interface. The CTI Server
translates and combines the various types of input data into a
unified, normalized format. Normalized format information is then
passed by the Message Router to various components of the system,
as required.
As noted above, the Voice Server in a specific embodiment has
several modules, including the Audio Recorder 145 and Call Record
Generator (CRG) 150. The Audio Recorder collects a plurality of
audio segments, representing the portions of a telephone call
during which the sound level exceeded an adjustable tolerance
threshold, thereby discerning alternating periods of speech and
silence. Functionally, the Call Record Generator (CRG) produces
Master Call Records, which encapsulate information (metadata)
describing a telephone call. This descriptive information comes
from a plurality of sources, including but not limited to an Audio
Recorder and a CTI Server. The call records are created using a
participant-oriented Call Record Model. The CRG then attempts to
match the call records with existing recorded audio data. The CRG
is thus able to combine data arriving in different chronological
order into a single manageable entity which describes the complete
history of a telephone call.
In a specific embodiment, a Playback Server (PBServer) (not shown)
is a sub-component within the Audio Recorder module which uses call
records to retrieve and play back telephone calls. Each recorder
has its own PBServer, which is connected to a Player module (not
shown) on the User Workstation 160. The Player module generally
contains a Stream Control Manager module, which enables the Player
module to use the PBServers to play back a telephone call which has
several different participants and thus may have portions of the
call stored on different recorders.
CTI Server
Still with reference to FIG. 1, when a call comes into the PBX
system, both SMDR and real-time CTI data are generated by the PBX,
and supplied to the recording system via the SMDR and CTI links. In
accordance with the present invention, these two types of data are
integrated by the CTI Server into a common format.
As known in the art, CTI (Computer Telephony Integration)
supplements the recorded audio data in several important ways. CTI
data is provided through a data communications link from specific
telephone switching equipment located at the customer site.
Supplied data comprises such items as the telephone numbers of
involved parties, caller ID/ANI information, DNIS information, and
agent ID numbers. ANI is Automatic Number Identification, a
signaling method that identifies the telephone number of the
calling party; the method is typically used by large-scale
commercial call centers. DNIS is Dialed Number Identification
Service, a feature that identifies the original "dialed digits,"
and that is commonly used in large-scale commercial call centers
when multiple directory numbers are routed to the same receiving
trunk group. In accordance with the present invention, the CTI
server performs the task of analyzing and reorganizing data from
both the real-time (CTI) and SMDR (asynchronous) links, and passing
the results onwards into the recording system for further
processing.
The design of the system of the preferred embodiment envisions that
there will be a number of CTI translation modules 165 to
accommodate a variety of possible input sources such as "native"
PBX interfaces, CTI "middleware" vendors, ISDN data channel
interfaces, etc. The system design incorporates flexibility in the
manner in which CTI information is collected, making the system
prepared to integrate with CTI links that may already exist at a
customer site. The CTI Server of the preferred embodiment is
capable of simultaneously monitoring both an SMDR link and a
real-time CTI link, gathering information about calls from both
sources, and combining that information into a single data model of
the telephony activity within the call center.
The CTI Server is responsible for supplying certain metadata
regarding telephony events to the Voice Server's Call Record
Generator 150. This metadata, such as called party and calling
party numbers, trunk and channel ID, date and time, agent ID, etc.,
is combined by the Call Record Generator along with the other
metadata, and data that is provided by the Audio Recorder 145
itself. Using this information, other components within the system
are able to search for calls using a wide variety of useful and
meaningful criteria, rather than simply using the recorder channel,
date and time. As is known to those skilled in the art, an "event"
is simply an action or occurrence detected by a computer program.
The Call Record Generator 150 integrates that data into a single
call record, which is updated after every event during the call, so
that at the end of the call, the call's entire history is contained
in the call record. The CRG matches the call record to the
recording segments created by the Audio Recorders. The CRG
integrates the call record with the metadata for the associated
recordings of the phone call to generate a Master Call Record. When
an operator wants to hear a recorded phone call, he uses the User
Workstation (preferably equipped with a graphical user interface)
to recall and play back the recorded call. Since the phone call may
have had several different participants, pieces of the call may
have been recorded on different recorders, each of which is
associated with a different Playback Server. The system is
nevertheless capable of playing back the entire phone call in the
proper sequence.
In a preferred embodiment the CTI Server obtains the information
regarding telephony events from various telephone switching
environments, including PBXs, ACDs, and turret systems, which may
have a wide variety of proprietary CTI interfaces. A telephone
switching environment is a local telephone system that provides for
routing of calls on a static or dynamic basis between specific
destinations; the system is capable of identifying of when calls
occur and who is involved in the calls. The CTI Server converts the
information received into a common "normalized" format that is a
simplified subset of the types of information available across the
different vendors' PBXs, ACDs, and turret systems. This data
conversion is partially facilitated by products such as Dialogic's
CT-Connect API, which is capable of processing CTI messages from
the major vendor's switches such as the Lucent DEFINITY G3, Nortel
Meridian and DMS-100, Aspect, Rolm 9751, Rockwell Spectrum and
Galaxy, Siemens Hicom, and Intecom. However, in accordance with the
preferred embodiment an additional software layer exists within the
CTI Server to further filter and normalize the CTI information.
This feature also allows for a separate point of integration with
customized software interfaces that may be necessary to connect
with other switch vendors, especially certain turret systems that
are not supported by Dialogic's CT-Connect (CTC) product. Alternate
embodiments of the translation module use Lucent CentreVu Computer
Telephony Server for Windows NT, or Genesys T-Server, as middleware
instead of Dialogic CT-Connect. Additional alternate embodiments
include direct "native" interfaces to a particular telephone
switch, such as Aspect, without an interposing middleware
product.
In terms of the CTI messages exchanged between the CTI Server and
the various PBXs, ACDs, and turret systems, in accordance with a
preferred embodiment the CTI Server is a "passive listener." That
is, the CTI Server will monitor and receive information about call
activity, but it will not send messages to affect, control, or
redirect the calls. Using an "active" CTI server is also
contemplated in specific embodiments.
Whereas the focal point of a Voice Server is recording content
(e.g., audio clips), the metadata generated by the CTI Server is
focused on describing the facts pertinent to the start and end
points of each participant's involvement within a call. In other
words, within the system of the preferred embodiment, recording is
managed in a call-centric (rather than event-centric) fashion. This
corresponds with the typical caller's point of view, in which a
call is the entire conversation with a business entity, even if the
conversation involves transfers to other agents or conferencing of
multiple parties. The CTI Server generates events with metadata for
the start and end points of the various recording segments of a
complex conversation. These event records are interrelated by ID
numbers and reason codes (see FIG. 3) so that the entire sequence
of events for a complex conversation can be reconstructed by a
browser application, preferably implemented on the User Workstation
160.
In accordance with the preferred embodiment, there can be one or
more CTI Servers within the system of the subject system, as needed
to process the traffic load of CTI information generated by
multiple PBXs, ACDs, and turret systems. In a specific embodiment,
a single CTI Server may be configured so as to connect with several
PBXs, ACDs, and turret systems, depending upon the traffic load and
physical connectivity requirements. In alternate embodiments,
different CTI servers can be attached to different input sources.
Generally, the number of CTI Servers within the system does not
have a direct relationship with the number of Voice Servers. The
telephony events generated by a CTI Server are individually
filtered and re-transmitted to the appropriate Voice Server based
upon configuration data for the system as a whole (managed by the
Central Database Server), which maps the recording locations
(extension number, or trunk & channel ID) with the Voice Server
name and recording input port (channel).
During the active lifetime of a call, real-time information is
accumulated within a historical call record that tracks each
participant within the call. Each participant record includes
descriptive fields for telephone numbers, agent ID numbers, time
ranges, and reason codes for joining and leaving the conversation.
At certain key points during the accumulation of data, whenever a
party joins or leaves the conversation, the call record is
transmitted onward to allow the rest of the recording system to
process the information accumulated thus far. Upon the conclusion
of the call, the CTI server retains a copy of the call record for a
configurable time interval before discarding it from memory. This
delay is intended to allow for the arrival of the SMDR data.
Upon receiving SMDR data, the CTI server searches its memory for a
call record pertaining to the same logical telephone call that
would have been accumulated from previous real-time messages.
Matching this information is not a trivial task, since the SMDR
link and real-time CTI link do not share a common reference ID
number for use by their messages in describing the occurrence of
the telephone call.
Therefore the software of the preferred embodiment must use other
"clues" to guide the matching process, by comparison on a
combination of other data fields that exists in common between the
SMDR and real-time CTI data. These data fields include: (1) the
telephone number of the first external party involved in the call;
(2) the telephone number of the first internal party involved in
the call; (3) the direction of the call (e.g., inbound, outbound);
(4) the start time of the call, in hours and minutes; and (5) the
duration, in seconds, of the call.
Once again, the matching process is not trivial because the SMDR
link gives the starting time of the call only in hours and minutes,
whereas the starting time given by the real-time link also includes
seconds. It is quite possible that more than one call could be
started and stopped within a single minute. This would result in an
ambiguous match, if not combined with other search fields. The same
argument holds true for each of the other fields upon which a match
can be performed. No single field alone will provide an unambiguous
matching of the records. Even in combination, it is conceivable
(although statistically unlikely) that an ambiguous case could
occur: if the same two parties were to call each other twice within
the span of a minute, and each call was roughly the same length in
seconds. The odds of such a problem are increased if a large number
of calls are routed through a common entry point into the call
center, as would be the case if the first internal party involved
in the call is a shared Voice Response Unit (VRU) or Automatic Call
Distribution (ACD) queue. In addition, if information about the
external party's number is missing due to limitations of the PSTN
or incoming trunk configuration, matching the call records becomes
even more problematic.
Adding to these difficulties is the fact that clock-time values
reported by the SMDR link and the real-time CTI link may not be
perfectly in synchronization with each other. Therefore, the
preferred embodiment comprises a mechanism in which an imperfect
match of times can be tolerated, while still retaining an
acceptable level of reliability in matching the call records.
Because these various factors require a degree of flexibility in
the matching algorithm, the preferred embodiment incorporates a
weighted formula that is applied to potential match candidates. The
formula yields a numerical confidence factor that can be used to
select the best apparent match candidate. For each of the "clues,"
a test is conducted to determine the quality of matching on that
data field. This matching quality is rated as a percentage. Certain
fields, such as time values, are allowed to vary within a
configurable tolerance range, whereas other fields are required to
match exactly or not at all. After the matching quality of a field
has been determined, it is multiplied by an importance factor that
applies a relative weight to each of the various fields that can be
examined during matching. The final confidence factor is the
summation of these calculations:
In order to account for the fact that characteristics of the call
traffic may vary significantly between individual call centers, the
tolerance factors (e.g., for time value offsets) and the weighting
factors are re-configurable. There is also a re-configurable
minimum level for confidence factors, below which the match
candidate will always be rejected.
For those fields, such as time or duration, where an imprecise
match may be allowed, the configuration data will define an
allowable variance range (plus or minus a certain number of
seconds). Values that do not match exactly, but fall within the
variance range, are rated with match quality expressed in
percentage that is measured by one minus the ratio of the
difference from the expected value versus the maximum variance.
Values outside the variance range are rated as a match quality of
zero. This produces a linearly scaled match quality. Alternate
embodiments may use other distributions (e.g., standard deviation
"bell curves") to produce a non-linear scale for the match quality.
Where an exact match is required for a field, the match quality is
either 100% or zero.
EXAMPLE
Real-time CTI events report a telephone call from an unknown
external party sing or deliberately suppressed ANI/CLID
information) to an internal party at extension 1234, starting at
12:25:03 and lasting for 17 seconds (CLID is Calling Line
Identification, a signaling method that identifies the telephone
number of the calling party; the method is typically used by
residential subscribers and small businesses). Two SMDR records
arrive which could possibly match with this call. The first record
indicates an inbound call received by extension 1234 at 12:26 and
lasting 26 seconds. The second record indicates an inbound received
by extension 1234 at 12:27 and lasting 20 seconds. The system is
configured with a variance range of plus or minus 3 minutes for the
start time, and plus or minus 10 seconds for the duration.
Weighting Factors are:
20 External Party Telephone Number 40 Internal Party Telephone
Number 30 Direction 20 Start Time 20 Duration
Confidence Factors are therefore calculated as follows:
system will therefore match the CTI events with the second SMDR
record.
After a match has been selected, the trunk channel information (and
any other useful information that can supplement the previously
gathered real-time CTI data) is extracted from the SMDR data and
added to the call record within the CTI server's data model of
telephony activity. Then the updated call record is transmitted
onward to allow the rest of the recording system to process it.
With the trunk channel information at hand, the recording system is
able to associate the enhanced logical search information with the
physical voice recording, and take whatever actions may have been
dependent upon this information, such as selectively recording or
discarding the call.
FIG. 2 is an illustration of the difference between trunk-side and
station-side recording at a call center with agents. With suitable
equipment for tapping into a voice communications line, a recording
unit can intercept telephone call traffic using either of these two
methods. By attaching wires for recording channels 180 on each
extension within a call center, the traffic can be intercepted and
recorded as it passes between the PBX 100 and the agent telephone
sets 230. This first method is known as "station-side" recording.
Alternatively, by attaching equipment 170 on the trunk lines
between the PBX and Public Switched Telephone Network (PSTN) 250,
the traffic can be intercepted at its point of entry into the call
center before the calls are dispatched by the PBX. This second
method is known as "trunk-side" recording. Since businesses usually
have more agent telephone sets than trunk lines, a "trunk-side"
solution is likely to require less recording equipment and thus be
less expensive. Another significant point for consideration is that
"trunk-side" provides access only to external inbound or outbound
calls, which are those typically involving customers of a business,
whereas "station-side" also provides access to internal calls
between agents (which may or may not relate to an external
customer's transaction).
A third type of recording interface is Service Observance 185 (see
FIG. 1), which is physically wired in manner like station-side
recording, but using separated dedicated lines to a recording input
channel rather than being interposed between a PBX and telephone
set, In this mode of operation, the Recorder joins into a telephone
call as a silent conference participant using the PBX Service
Observance feature (originally intended to enable a supervisor to
directly monitor an employee's telephone calls upon demand). This
differs from ordinary station-side recording in that the internal
party being recorded on a given input channel can vary upon demand
rather than being fixed by the wiring pattern.
FIG. 3 shows a line-chart that illustrates various parties involved
in a complex call. A is the customer phone number, and B and C are
the agent phone numbers located behind recording channels R20 and
R21 respectively (see FIG. 2).
Initially, the call comes in from line A 335 to line B 340. A
real-time CTI message occurs describing that phone B is ringing,
but not yet answered. B answers the phone 365 at time t0310. The
"NS" at 360 indicates the normal start of a phone call. A real-time
CTI message occurs describing the start of the call between A and
B. The telephony model is updated to reflect the fact that the call
between the initial 2 participants (A and B) started normally at
time t0310. A copy of the call record is then sent onward to the
rest of the recording system. The call record is retained within
the telephony model, associated with device (or line) B. At time
t1315, B places the call on hold 370 (the "XA" at 370 indicates
that the call was transferred away from B; the "XR" at 375
indicates that the transfer was received by HOLD). A real time CTI
message occurs describing that B placed the call on hold. The
telephony model is updated to reflect that B transferred the call
to HOLD 345 at time t1315. (This information is accumulated with
the information previously gathered at t0310). A copy of the call
record is then sent onward to the rest of the recording system. The
call record is removed from device B within the telephony model,
but kept in a list of held calls.
At time t2320, B returns to the call 380 and conferences in C 355
(the "XA" at 380 indicates that the call was transferred away from
HOLD; the "XR" at 382 indicates that the transfer was received by
B; the "CA" at 384 indicates that C was added as a conference
participant). A real-time CTI message occurs describing that B
returned to the call and invited C by conferencing. The call record
is moved within the telephony model from the list of held calls
back to device B. The telephony model is updated to reflect that
HOLD 345 transferred the call 380 back to B at t2320. (Note that
information is accumulated with the information previously gathered
at t0310 and t1315). A copy of the call record is then sent onward
to the rest of the recording system. The telephony model is updated
to reflect that C joined the call 384 as a conference participant
at t2. (This information continues to be accumulated with
previously gathered information). A copy of the call record is then
sent onward to the rest of the recording system. The call record is
retained with both devices B and C within the telephony model.
At time t3325, a real-time CTI message occurs describing that C
dropped out 386 of the call (the "CD" at 386 indicates that C was
dropped from the conference). The telephony model is updated to
reflect that C dropped out of the conference at t3. (This
information continues to be accumulated with previously gathered
information). A copy of the call record is sent onward to the rest
of the recording system. The call record is removed from device C
within the telephony model, but retained with device B.
At time t4330, A terminates the call to B. A real-time CTI message
occurs describing that A terminated the call (The "ND" at 390
indicates that a normal drop of the call occurred; the "OPH" at 395
indicates that the other party hung up). The telephony model is
updated to reflect that A stopped normally and B stopped because
the other party hung up at t4. (This information continues to be
accumulated with previously gathered information). A copy of the
call record is then sent onward to the rest of the recording
system. The call record is then removed from device B, but kept in
a list of completed calls. An SMDR message is received which
summarizes the call in its entirety. The list of completed calls is
searched to find a match, and the appropriate call record is
retrieved. The call record is updated with the trunk channel
information from the SMDR message. A copy of the call record is
sent onward to the rest of the recording system. The call record is
removed from the list of completed calls.
FIG. 4 shows a schematic block diagram of a preferred embodiment
for translating, summarizing and normalizing signals received from
both an SMDR link and a Dialogic CT-Connect CTI unit. In the
embodiment illustrated in FIG. 4, the recording system of the
subject system is represented by daVinci.TM., a new generation
recording system of Dictaphone Corp. Alternatively (or
simultaneously), Dictaphone's Symphony.TM. CTI software can be
used, in conjunction with Dictaphone's ProLog.TM. recording system
(the system preceding daVinci.TM.). Hereinafter, the
translation/summarization module of the preferred embodiment
illustrated in FIG. 4 will be referred to as CtiCtc.exe.
The module CtiCtc.exe is itself comprised of a plurality of
modules, as shown in FIG. 4. A CtiAgentEvent module 448 is
comprised of a data structure for agent log-on and log-off
messages. A CtiAgentStatusFile module 454 manages a file that
tracks agents currently logged on. A CtiCallEvent module 416 is
comprised of a data structure for a call record (i.e., normalized
and summarized CTI events). A CtiCallState module 418 is comprised
of a generic data structure to represent the state of telephony
activity at a particular location (extension, hold area, etc.). A
CtiComMessageEmitter module 476 comprises a layer that converts the
stream of CtiCallEvent objects (generated by a CtiCtcAnalyzer 456)
into a format that can be sent to other da Vinci system components.
A CtiCtcAnalyzer module 456 comprises a processing engine which
examines CTC and SMDR messages and keeps track of a state machine
for the activity on each extension. The CtiCtcAnalayzer module
performs normalization of the CTC and SMDR data.
A CtiCtcAnalyzerUtils module 452 comprises a collection of utility
subroutines that assist in examining the CTC and SMDR messages. A
CtiCtcCallState module 420 comprises a data structure that
represents the state of telephony activity at a particular location
(extension, hold area, etc.) including CTC-specific information. A
CtiCtcCallStateList module 432 manages an open-ended collection of
CtiCtcCallState objects. This collection of objects is typically
used to track calls that are "held" or "bumped." A CtiCtcData
module 428 comprises a data structure wrapped around the raw CTC
data, with the addition of a time stamp indicating when a message
arrives. A CtiCtcDataFile module 412 manages a file of CtiCtcData
objects that can be captured or displayed. A CtiCtcExtensionInfo
module 442 manages a collection of CtiCtcCallState objects, with
one object for each extension.
A CtiCtcInput module 464 comprises an input source engine that
obtains incoming CtiCtcData objects, either from a "live" server or
from a playback file. A CtiCtcMain module comprises the main( )
function for CtiCtc.exe. The main( ) function handles command line
and registry parameters, along with other start-up processing. A
CtiCtcParameters module 472 comprises data structure and program
logic for managing the configuration parameters in the Windows NT
registry. A CtiCtcScanner module 446 comprises a utility module for
building a list of all available extensions on a particular
telephone switch. A CtiCtcStats module 434 comprises a data
structure for compiling statistics on the number of CTC, SMDR, and
CTI messages. A CtiDtpField module (not shown) is used by a
CtiDtpMessageEmitter module 478, and comprises a data structure for
an individual field in the Dictaphone Telephony Protocol ("DTP"),
used to communicate with other Symphony CTI system components. A
CtiDtpMessage module (not shown) is used by a CtiDtpMessageEmitter
module 478, and comprises a data structure for a complete message
in the DTP to be sent onwards to the Symphony CTI system.
A CtiDtpMessageEmitter module 478 comprises a layer that converts
the stream of CtiCallEvent objects (generated by CtiCtcAnalyzer
456) into a format that can be sent to the Symphony CTI recording
platform. A CtiDtpSocketSrv module (not shown) manages the TCP/IP
connection through which messages for DTP are sent to the Symphony
CTI platform. A CtiDtpUtility module (not shown) comprises a
collection of utility routines that assist in examining and
processing DTP messages. A CtiExtensionFile module 450 manages the
configuration file that lists all available telephone extensions. A
CtiExtensionInfo module 440 manages a collection of CtiCallState
objects, with one object for each extension. A CtiExtensionNumber
module 430 comprises an abstraction of an individual extension
number as either a numerical or string value, so that changes to
this model will not have a global impact in CtiCtc.exe.
A CtiMessageEmitter module 458 comprises an abstract layer that
converts the stream of CtiCallEvent objects (generated by
CtiCtcAnalyzer 456) into a format that can be sent to various
target platforms, including the da Vinci and SymphonyCTI systems. A
CtiMessageEmitterParameters module 474 comprises a data structure
and program logic for managing configuration parameters that relate
only to the message emitter(s). A CtiMessageQueue module 462
comprises shared memory for transferring data between a threads. As
is known to those skilled in the art, a "thread" is a part of a
program that can execute independently of other parts. A
CtiNulMessageEmitter module 460 comprises a layer that accepts the
stream of CtiCallEvent objects (generated by CtiCtcAnalyzer 456)
and discards them instead of sending them to a target platform.
Typically this layer is used only when debugging CtiCtc.exe, or to
capture a sample file of CTI events from a PBX without sending them
to the da Vinci or SymphonyCTI systems. A CtiPartyListElement
module 414 comprises a sub-component of the CtiCallEvent data
structure 416. The module 414 tracks information about an
individual participant (e.g., caller, recipient) in a call.
A CtiPeriodicMsg module 468 comprises a generic handler for sending
timer-based housekeeping messages. A CtiPrint module 444 comprises
a layer that manages console output and conditional trace messages.
A CtiSmdrData module 424 comprises a data structure wrapped around
the raw SMDR data, with the addition of a time stamp indicating
when a message arrives. A CtiSmdrDataFile module 408 manages a file
of CtiSmdrData objects that can be captured or replayed. A
CtiSmdrDataList module 422 manages an open-ended collection of
CtiSmdrData objects. This is typically used to buffer SMDR records
that have not been paired with CTC records. A CtiSmdrInput module
466 comprises an input source engine that obtains incoming
CtiSmdrData objects, either from a "live" server or from a playback
file.
A CtiTagNames module 436 comprises a utility module that converts
number values to descriptive strings for debugging and tracing
purposes. A CtiTime module 438 comprises a utility module that
converts time values to UTC for internal storage and conditionally
prints times in either the UTC or local time zone. A CtiTrunkMap
module 426 comprises a data structure that describes a mapping
between logical trunks and logical trunk groups, into physical
trunks and TDM timeslots. A CtiTrunkMapFile module 410 manages a
configuration file that contains the CtiTrunkMap information.
FIG. 5 illustrates the steps by which the translation module
CtiCtc.exe integrates the data received from the CTI and SMDR
links. Initially, at step 502, the translation module receives a
message from the SMDR link or from the CTI link. If the message is
determined, at step 504, to be a CTI message, the current data
model of telephony activity is updated at step 506. If the
translation module determines at step 514 that the CTI message
indicates a party joined or left the call, the call record is at
step 518 transmitted onward to the rest of the recording system
before continuing to step 512. Otherwise, no message is transmitted
onward to the rest of the recording system and processing continues
directly to step 512. If the translation module determines at step
512 that the CTI message indicates that the call has been
concluded, at step 520 the module removes the call record from the
associated devices. The translation module then adds the call
record to the list of recently completed calls at step 528.
Completed calls are discarded (step 530) after they get too old
(i.e., after a predetermined number of recorded calls, or a given
time period after the original recording of the call). Processing
then continues again from step 502 by receiving the next incoming
message. If at step 512 the call has not been concluded, the
completed calls are discarded (step 530) after they get too old.
Processing then continues again from step 502 by receiving the next
incoming message.
If at step 504 the message is an SMDR message, the translation
module at step 508 scans the list of recently completed calls. At
step 510 the translation module calculates confidence factors for
the recently completed calls by using the formula:
Confidence Factor=.SIGMA..sub.i ((Match Quality).sub.i *(Weighting
Factor).sub.i)
If any matches are found (step 516), and more than one match is
found (step 522), the match with the highest confidence factor is
used (step 526). If only one match is found, that match is used
(step 524). At step 540, the trunk channel information is
extracted, and at step 544 the call record is updated within the
list of recently completed calls. The call record is transmitted at
step 548 to the rest of the recording system. At step 550 the call
record is discarded from the list of recently completed calls.
Cpleted call are discarded (step 530) after they get too old. If no
matches were found at step 516, the completed calls are discarded
(step 530) after they get too old. Processing then continues again
from step 502 by receiving the next incoming message.
As shown in FIG. 6, the CTI Server can be viewed as a set of
logically distinct layers that deal with translating and
distributing CTI events. Starting from the bottom of the picture,
CTI events flow from a PBX in its proprietary format to Dialogic
CT-Connect middleware 640 another API layer 650 or custom interface
layer 660 that each provide partial normalization of the data. This
helps to reduce the complexity of the "translation" job, since
there are fewer APIs than individual PBX types. But since one
object of the subject system is to retain the flexibility to
integrate with a variety of third-party CTI vendors (e.g.,
Dialogic, Genesys, etc.) there is another layer 670 above the API
or custom interface layer to complete the job of "translation." The
final result after passing through this "normalization" layer is
that all of the CTI events are in a single, common, integrated data
format.
Once the CTI events have been converted to a normalized format, the
CTI Server can address its other mission of distributing (routing)
the messages. The distribution layer 680 examines each message to
determine what other recording system components need to receive
it, and then sends a copy of the event to the appropriate
destination(s).
This logical separation of responsibilities used in a preferred
embodiment simplifies the programming required to implement the
subject system. Translation modules do not need to know anything
about other recording system components, and they can focus on
dealing with a single specific PBX or vendor API layer. Likewise,
the distribution module will not need to know anything about
specific PBX or vendor API layers, and it can focus on making
routing decisions and communicating with the rest of the recording
system.
FIG. 7 illustrates how, in addition to telephony events, the CTI
Server 710 used in accordance with the present invention is
responsible for supplying certain metadata regarding agent events
to the System Controller, which is part of the Scheduling &
Control Services 130 shown in FIG. 1. This information, which
generally includes agent ID, extension number, logon and logoff
time, etc., is obtained when available from the various PBXs, ACDs,
and turret systems. The agent events delivered to the System
Controller 130 enable a map to be maintained of the extension
number(s) where a real person can be found, at a given date and
time. This information enables a browser application to
intelligently associate some of the previously recorded calls even
if a person was using different telephone sets according to a `free
seating` plan. The CTI Server 710 also keeps a local cache of the
agent information, so that agent information can be included when
sending the telephony events to the Call Record Generator 150.
The physical layout of the CTI Server used in a specific embodiment
is shown in FIG. 8. With reference to FIG. 1, the translation
modules are implemented by separate programs, such as CtiCtc.exe
406, which encapsulate the details on converting a specific PBX
interface or vendor API layer into a normalized format. The
distribution module is preferably implemented by a single program,
CtiServ.exe 820, which includes the main processing and routing
logic for the CTI Server.
As noted, the translation modules of the CTI Server convert
proprietary-format CTI information into a normalized format. In
accordance with a preferred embodiment, this is done in several
layers within the program. The information is first converted by
Dialogic's CT-Connect software into the CTC-API format, and then
the conversion to the generic format used by the other components
of the recording system is completed by the translation module
CtiCtc.exe. Once the data is converted, it is transmitted to the
distribution module (CtiServ.exe) by using a distributed
communications method such as DCOM. Component Object Model (COM) is
a Microsoft specification that defines the interaction between
objects in the Windows environment. DCOM (Distributed Component
Object Model) is the network version of COM that allows objects
running on different computers attached to a network to interact.
An alternate embodiment of the CTI Server utilized Microsoft
Message Queue (MSMQ) technology as the means to carry messages
among the system components, instead of the original DCOM method
used by CtiServ.exe, and those skilled in the art would appreciate
that a variety of additional data communications technologies are
also suitable to this role.
The translation module and the distribution module of the CTI
Server can be located on different machines, if desired. There can
be multiple translation modules running in the system--one for each
PBX or CTI middleware environment. There can also be different
types of translation modules, with one version for each interface
or API layer. As depicted in FIG. 8, CtiCtc.exe deals with the
Dialogic CT-Connect API, and there are 3 copies of this program
running to handle the PBXs. If other types of APIs are used, there
would be other programs for these various interfaces. All
translation modules contribute data upward to the distribution
module in a single, common, normalized format. An example of a
version of CtiCtc.exe configured to work with a Lucent Telephony
Services interface (and thus called CtiLts.exe instead of
CtiCtc.exe) is shown in FIG. 9. The modules which are common to
both versions of the program are shown in FIG. 9 as shaded gray.
The unshaded modules represent those portions of the program that
necessarily vary between CtiCtc.exe and CtiLts.exe, due to the
differing input parameters and data structures used by both
systems.
Again with reference to FIG. 8, the distribution module
(CtiServ.exe) receives and collects all the CTI events from the
various translation modules. Then it puts the events into a single
inbound queue 830 for processing by a main control thread 835.
After the events are processed, they are separated into individual
outbound queues 840. Finally, the events are sent by various
delivery threads 850 to the CRG components within different Voice
Servers. The main processing thread 855 (WinMain) is deliberately
isolated (decoupled) from the inputs and outputs to ensure that
delays in transmitting or receiving data will not impact the
overall performance of the CTI Server.
FIG. 11 shows how the CTI server in accordance with a specific
embodiment consists of several threads that implement three
distinct layers of processing (data collection 1110, data
normalization 1120, and message emission 1130). FIG. 12 illustrates
the processing steps of these layers. The dashed lines indicate
message flow between threads, whereas the solid lines indicate
program logic flow. The CTI translation modules are thus internally
separated into 3 major sub-tasks; (1) data collection from the
input source (PBX, CTI middleware, etc.); (2) normalization of the
data to a common format; and (3) communications with the system
platform.
In a data collection layer, the initial step 1210 is to open the
connection to the CTI data source. At step 1214 the layer receives
a CTI event, and at step 1216 posts the CTI event to the Message
Queue 462 (see FIG. 4). If at step 1218 a shutdown is in progress,
the connection to the CTI data source is closed at step 1220, and
at step 1222 data collection is ended. If at step 1218 a shutdown
is not in progress, the CTI connection remains open (step
1212).
At step 1228, the data normalization layer receives a CTI event
from the Message Queue 462. The data normalization layer updates
the telephony model at step 1230. See FIG. 13 for a more detailed
explanation of the updating of the telephony model. At step 1231,
the call state is posted to the Message Queue, if necessary. At
step 1232 completed calls are discarded from memory after they age
beyond a configurable time limit. At step 1233 the "hang-up"
routine is called to update the telephony model for held or bumped
calls after they age beyond a configurable time limit. At step
1234, if a shutdown is in progress, the data normalization layer
checks the inbound message queue at step 1236. If the message queue
is empty, data normalization is ended (step 1238). If the message
queue is not empty at step 1236 or if there is not a shutdown in
progress at step 1234, the data normalization layer goes to step
1226 and waits for the next CTI event to arrive.
The message emission process begins with opening a connection to a
target platform, such as the da Vinci or SymphonyCTI recording
systems at step 1240. At step 1244, the message emission layer
receives the call state from the message queue 462. At step 1246,
the call state data is converted into a platform-specific format.
At step 1248, the message emitter sends the message to the target
platform. At step 1250, if a shutdown is in progress, a check is
made at step 1252 for whether the inbound message queue is empty.
If the inbound message queue is empty, message emission is ended at
step 1254. If the inbound message queue is not empty at step 1252,
or if there is not a shutdown in progress at step 1250, the message
emission layer, at step 1242, maintains the open connection to the
target platform and awaits the next call state transmission.
Master Call Record
The CTI Server sends "Call Event Records" onward to the recording
platform. These messages provide details on the start and end of
calls, as well as significant transitions that affect the lists of
participants for the calls. The list of participants is cumulative,
and information regarding participants is retained for the entire
duration of the call even when some participants in the list may
have dropped off from the call. If a participant rejoins the call,
a new, separate entry will be created to reflect that change within
the participant list. The following table shows the fields
contained within these messages.
CtiCallEvent Type Name (max length) Description Version WORD
Version number of this message format, for reverse compatibility.
MessageID GUID Unique ID for this message instance RecorderNode
WORD A number that identifies a particular Voice Server
RecorderChannel WORD A number that identifies a recording input
channel on a Voice Server EventType BYTE Indicates if this event
added (0x01) and/or dropped (0x02) participants in the call.
EventReason BYTE Indicates if this call was affected by a normal
(1), conference (2), or transfer (3) telephony event CTICallRecId
GUID Unique ID pertaining to entire call (CTI server provides the
same ID for a call that is transferred, conferenced, etc)
CallDirection BYTE Indicates call origin - outbound (0x12), inbound
(0x21), internal (0x11), or unknown (0x44) RingLength WORD Seconds
between the first ring signal and going off-hook (picking up the
phone) DTMFCode String*(50) DTMF codes entered during the call
ApplicationData String*(32) Character array dedicated to
information the switch may provide along with the call (e.g.,
account number) CallingParty WORD Index number of the calling party
within the participant list. Normally this is zero. CalledParty
WORD Index number of the called party within the participant list.
Normally this is one. PBXCallRecld DWORD Number provided by the PBX
to identify this call. NumberOf- WORD Count of participants in the
following Participants array. ParticipantList Vector* Array of
PartyListElement describing all participants involved in the call
*ObjectSpace data types
ObjectSpace is a set of C++ class libraries provided by
ObjectSpace, Inc., that define useful general-purpose data
structures including representations of strings, time values, and
collections of objects (such as vector arrays or linked lists).
These class libraries are implemented in a way that supports a wide
variety of computer operating systems. Those skilled in the art
will appreciate that many alternate implementations for such data
structures are suitable for this role.
CtiPartyListElement Name Type Description AgentID String*(24)
Registered ID of person, typically used for "free seating" call
center environments Number String*(24) Telephone number of this
participant (e.g., ANI, DNIS, Dialed Digits) Consol String*(10)
Seating position that can consist of one or more stations. Station
String*(10) Unique telephone set, possibly with multiple
extensions. Extension String*(6) Internal line number of the
participant SwitchId WORD Number of the switch (PBX, ACD, or turret
system) which is handling the conversation TrunkID WORD
Identification of trunk line which is handling the conversation
VirtChannel WORD Identification of trunk's channel (time slot)
which is handling the conversation Location- BYTE Describes the
location of Reference participant with respect to the switch - can
be internal (1), external (2), or unknown (3) StartTime
time_and_date* Time participant joined the call EndTime
time_and_date* Time participant left the call ConnectReason BYTE
How participant joined the call: norm start of call (1), being
added to a conference (2), or receiving a transferred call (3)
Disconnect- BYTE How participant left the call: Reason normal end
of the call by hanging up (1), dropping out of a conference (2),
transferring away a call (3), or call ends by another party hanging
up (4). Changed BOOL Indicates if recent change in CTI message:
*ObjectSpace data types
For external participants, only the fields Number, SwitchName,
TrunkID, VirtChannel, LocationReference, StartTime, EndTime,
ConnectReason, and DisconnectReason will be applicable. For
internal participants, all fields may be applicable. Unused string
fields will be null terminated. Unused number fields are set to
zero. Each call event record will contain at least two participants
in the list. These two participants are the original calling party
(0) and called party (1) and will appear within the list in that
order respectively.
Note: The data field "Number" will be filled in a variety of ways,
depending upon the type of participant and direction of the
call.
Participant Type Call Direction Number Field External participant
Inbound Call ANI External participant Outbound Call Dialed Digits
Internal participant Inbound Call DNIS or Extension Internal
participant Outbound Call Extension Internal participant Internal
Call Dialed Digits or Extension
The CTI Server sends "Agent Event Records" onward to the recording
platform's System Controller to convey information when an agent
logs on/off at a particular location. The following table shows the
fields contained within these messages.
CtiAgentEvent Type Name (max length) Description Version WORD
Version number of this message format, for reverse compatibility.
MessageID GUID Unique ID for this message instance EventType BYTE
Indicates if this event pertains to either a logon (1) or logoff
(2). LocationType BYTE Indicates if this event pertains to a
location type such as a console (1), station (2) or extension (3).
AgentID String*(24) Registered ID of person, typically used for
"free seating" call center environments SwitchID WORD Number of the
switch (PBX, ACD, or turret system) where the agent connected.
Console String*(10) Seating position that can consist of one or
more stations. Station String*(10) Unique telephone set, possibly
with multiple extensions. Extension String*(6) Internal line number
of the participant StartTime time_and_date* Time that the agent
logged in. EndTime time_and_date* Time that the agent logged out.
*ObjectSpace data types
Within any given "Agent Event Record", only one of the following
three fields will be applicable: Console, Station, or Extension.
The actual mapping is determined by the LocationType. Unused string
fields will be null terminated. Unused number fields are set to
zero.
It will be appreciated that the general principles behind the
method described above are suitable not only for associating and
combining real-time CTI data with the trunk channel information
from an SMDR message, but also for any situation where a mixture of
information is being provided from two or more sources and there is
a need to gather and merge the information to get a more complete
picture of what is actually happening in the system. The disclosed
method could easily be adapted by those of ordinary skill in the
art to situations in which the mapping or association between the
multiple sources of information is "weak" and prone to ambiguity.
While this method does not make the potential ambiguity disappear,
it helps to define a quantitative set of rules for making a
judgement call on when a match is "good enough" to act upon. While
human beings are often capable of making such judgement calls
intuitively, computers need a specific set of instructions in order
to act in a repeatable and reliable fashion upon the input
data.
Previous recording systems that made use of CTI to collect enhanced
search information mimicked the event-oriented interfaces provided
on the data links from a PBX. Individual database records were
constructed on a 1-to-1 basis for the events occurring during the
total lifetime of a phone call. The interpretation of the series of
events was left to the end user. Associations between related
events were made difficult in certain cases because the call
identification numbers given by a PBX may change after a call has
been transferred or conferenced, or the numbers may be recycled and
reused over time. Following and tracing the history of events for a
complete call from the perspective of the external customer could
require much manual and repetitive searching. Playing back the
entire set of audio recordings from the start of that customer's
interaction with the business, to the ultimate conclusion of that
customer's transaction, could also require additional repetitive
manual requests to play back the individual recorded segments
within a call that was transferred or conferenced.
To resolve this problem, the CTI server of the preferred embodiment
maintains and accumulates information within a data model of
telephony activity. FIG. 10 depicts the key elements of the data
model. This consolidated information is shared with the rest of the
recording system when parties join or leave a call, thereby
eliminating the need for downstream components to store or
interpret the individual CTI events occurring during a call's
lifetime.
During the active lifetime of a call, real-time information is
accumulated within a historical call record that tracks each
participant within the call. At certain key points during the
accumulation of data, whenever a party joins or leaves the
conversation, the call record is transmitted onward to allow the
rest of the recording system to process the information accumulated
to that point. Upon the conclusion of the call, the CTI server of
the preferred embodiment retains a copy of the call record for a
configurable time interval before discarding it from memory. This
delay allows for the arrival of the SMDR data.
The call records are organized into a two-tiered hierarchy of calls
and participants. Certain data fields that apply globally to the
entire call are stored at the upper level. Most data fields,
however, apply only to a specific party involved within a call, and
are stored at the lower level. Individual participants can have
identifying information (such as extension number, agent ID,
telephone number via DNIS/ANI/CLID, trunk and channel) along with
time-stamps and reason codes for the entry and exit from
participation in the telephone call. Reason codes include initial
start, transfer, hold, resume, conference add/drop, and
hang-up.
The currently active call on each telephone set being monitored is
maintained within a storage area 1020 of the data model. Also, the
data model provides for an open-ended list 1040 of calls that may
be "on hold" (and therefore not associated with any telephone set).
There is also a list 1030 that can be used temporarily for calls
when they are in a state of transition during transfers, queuing or
re-routing, for the brief period of time when an active call is
disassociated from its original telephone set but not yet
associated with a new telephone set. Finally, there is a list 1050
of recently completed calls that is used to await additional
information that might be provided from a SMDR message.
This complete set of data structures is replicated independently
for each CTI server that monitors a separate PBX within the overall
call center environment.
The call-centric structure and the list of participants facilitate
a common framework for modeling the various types of complex call
scenarios that may occur during the life of a call, far beyond the
simplest example of a basic two-party telephone call. Moreover, the
recording units can link references (i.e., logical pointers) to the
audio recordings for a portion of the call, so that these audio
sections are associated with the total history of the logical
telephone call. Each call record can be linked within the database
to an open-ended list of references, which provides: the name of a
Voice Server; the name of a .WAV file containing the audio
recording; the offset within the .WAV file to the start of the
recording segment; the start time of the recording segment; and the
duration of the recording segment.
Rather than relying exclusively upon the call identification number
assigned by the PBX, the CTI server of the preferred embodiment
obtains a Globally Unique Identifier (GUID), that is generated at
the software's request by the underlying Microsoft Windows NT
operating system, and uses that GUID to identify the call uniquely
within the recording system's memory, online storage database, and
offline storage archives. The GUID is initially requested at the
start of the call. While the call remains active, the CTI server
maintains a record of both the call identification number assigned
by the PBX, and the GUID assigned to the call by the software of
the preferred embodiment. When a CTI event arrives, the system
searches the telephony model to find a matching call record for the
PBX-assigned call identification number. At transition points
during a call's lifetime, such as when it is transferred or
conferenced, the PBX typically provides the old and new
identification numbers together in that single transition event. In
these cases, after locating the matching call record, the software
of the preferred embodiment updates its record of the call
identification number now being used by the PBX while retaining the
originally allocated GUID value. In this way, the same GUID
identifies the call throughout its lifetime, even while the PBX
call identifier may be changing. The long-term uniqueness of the
GUID value is also useful if the PBX recycles and reuses previously
assigned call identifiers. It further helps in dealing with calls
within a multiple PBX environment. While another PBX may
coincidentally use the same call identification number, a different
GUID is assigned at the start of each individual call, thereby
avoiding a conflict within the telephony model.
As shown in FIG. 11, the CTI server consists of three distinct
layers. Each layer actually runs in a separate thread of execution,
and communicates with the other layers through shared memory,
control semaphores, and message queues. The first layer 1110 is
responsible for gathering input from the PBX data link(s), and
there can actually be several threads running to provide better
throughput capacity or to handle multiple diverse input sources
(e.g., SMDR and real-time CTI messages). After saving the clock
time when a message is received, the first layer 1110 places the
message into a queue for subsequent processing by the second
"analyzer" layer. The second layer 1120 is responsible for updating
and maintaining the telephony model within the memory of the CTI
server, and for deciding when to send copies of call records onward
to the rest of the recording system. When a call record needs to be
sent onward, the call record is placed into a message queue for
subsequent processing by a third "message emitter" layer 1130,
which is responsible for communications with other components of
the overall recording system. This separation of layers gives the
CTI server the flexibility to process its input and output sources
in a de-coupled fashion, so that any delay in one area of
communications does not affect the processing of another area. In a
sense, the design approach provides a virtual "shock absorber" so
that bursts of input traffic, or temporary lag times in
communicating with other parts of the recording system, can be
tolerated without loss of data or incorrect operation of the
system.
The call records saved within the telephony model also include a
record of the last state of the device as reported by the PBX. This
information is used by the analyzer to run state machine rules, in
order to select a handler routine for a subsequent message. The CTI
server uses the previous state of the device (e.g., ringing,
answered, and so forth) along with the current state of the device
to select a handler routine from a matrix of potential choices.
The analyzer layer is of particular interest, since it is
responsible for updating and maintaining the data model of
telephony activity. Its overall program logic flow is illustrated
in FIG. 12 and the subroutine called at step 1230 is shown in
further detail by FIG. 13. This program logic is described below.
1. Receive a CTI event from the message queue at step 1228. 2.
Enter the subroutine at step 1230 to update the telephony model.
Referring now to FIG. 13, search the data of model of telephony
activity, to find a matching record at step 1322 with the same
monitored device (i.e., telephone set). 3. If the PBX-assigned call
identification number does not agree, search for a matching record
in the lists of calls on hold, in transition states, or recently
completed. If a match is then found, move the call record on the
affected device to the list of calls in transition states, and move
the matching record to the monitored device. 4. At step 1324, use
the previous state as recording within the telephony model, along
with the new state reported in the CTI event, to select the
appropriate handler routine at step 1332 from a matrix of choices.
The handler routine will be one such as those described below. 5.
At step 1340, run the steps of the handler routine. This will
commonly include steps to save at step 1342 information from the
CTI event into the call record, to update the call-related portion
of the Object Status, if necessary (step 1344), to update
Participants within the Object Status, if necessary (step 1352), to
run additional action methods or handler routines for other
affected telephony objects, if necessary (step 1348), and to post
Object Status to the message Queue for the Emitter to a target
platform (step 1354). 6. At step 1360, returning to FIG. 12, at
step 1232, discard completed calls within the data model of
telephony activity, if they have aged beyond a certain
re-configurable time limit. 7. Call the "hang-up" routine at step
1233 for any held call that have aged beyond a separate
re-configurable time limit. Likewise, call the "hang-up" routine
for any calls marked in transition, which have aged beyond another
separate re-configurable time limit. 8. Continue again from the
beginning of this logical program flow at step 1226.
The following description lists processing steps for various
handler routines that may be called in response to certain event
types using a decision matrix based upon past and current state
information.
Handler Routines Ignore: adjust state based on CTI event DialTone:
save the initial start-time of the call save the original dialed
number, if available adjust state based on CTI event RingIn: adjust
state based on CTI event time-stamp when ring occurred clear call
record set inbound, outbound, internal Answer: adjust state based
on CTI event compute total ringing duration fill in call record
with calling party & called party generate START message to
recording system Abort: adjust state based on CTI event clear
timers & original dialed number Hang-Up: adjust state based on
CTI event update call record to stop all parties indicate which
party actually hung up on the call generate STOP message to
recording system RingOut: adjust state based on CTI event
time-stamp when ring occurred (i.e., now) clear call record set
inbound, outbound, internal compute total ringing duration (i.e.,
zero) fill in call record with calling party & called party
generate START message to recording system Hold: adjust state based
on CTI event stop participant placing the call on hold add new
placeholder participant for HOLD generate TRANSFER message to
recording system move call record to hold area fill device slot
with a new empty call record Resume: if device slot not idle, move
call record to transition list move matching call record from hold
area to device slot adjust state to "active" stop the placeholder
participant for HOLD add new participant for telephone set that
resumes the call generate TRANSFER message to recording system
Conference: if call record found in hold area, if device slot not
idle, move call record to transition list move matching call record
from hold area to device slot adjust state to "active" stop the
placeholder participant for HOLD add new participant for telephone
set that resumes the call generate TRANSFER message to recording
system adjust state based on CTI event add new participant for
telephone set that is added via conference generate CONFERENCE-ADD
message to recording system Transfer: if call record found in hold
area, if device slot not idle, move call record to transition list
move matching call record from hold area to device slot adjust
state based on CTI event stop the participant leaving the scope of
the call (either a device or HOLD) add new participant receiving
the transferred call generate TRANSFER message to recording system
ConfDrop: adjust state based on CTI event stop the participant
leaving the scope of the call generate CONFERENCE-DROP message to
recording system OpAnswer: adjust state based on CTI event
re-compute total ringing duration correct the affected participant
entry in the call record generate CORRECTED message DestChanged:
clear call record the call will be processed via a subsequent CTI
event
The following step-by-step description describes the same call
scenario as in FIG. 3, but with emphasis on the data model of
telephony activity. 1. A real-time CTI message occurs describing
that phone B is ringing, but not yet answered. 2. The "Ringin"
routine is invoked. 3. The telephony model is updated with the time
when ringing started (for use later in measuring ring duration) and
the call direction. These facts are stored with device B 340. 4. A
real-time CTI message occurs describing the start of the call
between A 335 and B 340. 5. The "Answer" routine is invoked. 6. The
telephony model is updated to reflect the initial 2 participants (A
and B) started normally at t0310. 7. A copy of the call record is
sent onward to the rest of the recording system. 8. The call record
is retained within the telephony model, associated with device B
340. 9. A real time CTI message occurs describing that B 340 placed
the call on hold. 10. The "Hold" routine is invoked. 11. The
telephony model is updated to reflect that B 340 transferred the
call to HOLD 345 at t1315. (This information is accumulated with
the information previously gathered at t0). 12. A copy of the call
record is sent onward to the rest of the recording system. 13. The
call record is removed from device B 340 within the telephony
model, but kept in a list of held calls. 14. A real-time CTI
message occurs describing that B 350 returned to the call and
invited C 355 by conferencing. 15. The "Conference" routine is
invoked. 16. The call record is moved within the telephony model
from the list of held calls back to device B 350. 17. The telephony
model is updated to reflect that HOLD 345 transferred the call back
to B 350 at t2320. (Note that information is accumulated with the
information previously gathered at t0 and t1). 18. A copy of the
call record is sent onward to the rest of the recording system. 19.
The telephony model is updated to reflect that C 355 joined the
call as a conference participant at t2320. (This information
continues to be accumulated with previously gathered information).
20. A copy of the call record is sent onward to the rest of the
recording system. 21. The call record is retained with both devices
B 350 and C 355 within the telephony model. 22. A real-time CTI
message occurs describing that C 355 dropped out of the call. 23.
The "ConfDrop" routine 386 is invoked. 24. The telephony model is
updated to reflect that C dropped out of the conference at t3.
(This information continues to be accumulated with previously
gathered information). 25. A copy of the call record is sent onward
to the rest of the recording system. 26. The call record is removed
from device C within the telephony model, but retained with device
B. 27. A real-time CTI message occurs describing that A terminated
the call. 28. The "Hang-Up" routine is invoked. 29. The telephony
model is updated to reflect that A stopped normally and B stopped
because the other party hung up at t4330. (This information
continues to be accumulated with previously gathered information).
30. A copy of the call record is sent onward to the rest of the
recording system. 31. The call record is removed form device B 350,
but kept in a list of completed calls. 32. A SMDR message occurs
summarizing the call in its entirety. 33. The list of completed
calls is searched to find a match, and the appropriate call record
is retrieved. 34. The call record is updated with the trunk channel
information from the SMDR message. 35. A copy of the call record is
sent onward to the rest of the recording system. 36. The call
record is removed from the list of completed calls.
FIG. 14 depicts the flow of information within the remainder of the
recording system. The same enhanced search information S11412 is
provided by the CTI server to all of the recording units involved
in handling a portion of the call. Even if a call is transferred to
another telephone set, which is attached to an input channel on a
different recorder, the entire call will still remain associated as
one entity within the system. Each recorder maintains a local copy
of the audio sections V11416, V21420, and V31424 that it obtained
during the call, along with a complete call record containing
search information S11412 which contains the two-tiered call and
participant model. The search information is copied to a central
database server 1450, along with references (i.e., logical
pointers) to the original audio recordings VR11428, VR21432, and
VR31436. When a user searches for a call, the search results 1465
will include the complete call record S11412. By using the audio
references the playback software can reassemble the complete audio
for the original call, including sections possibly obtained from
different physical recording units.
The general principles behind the method described above would be
suitable, not only for representing the complete history of
telephone call's lifetime, but other forms of multi-party
communications. This may include certain forms of radio traffic
that have an associated data link, which provides "talk group"
identification numbers (or similar types of descriptive search data
in relation to the audio traffic).
Call Recorder Generator
The Call Record Generator (CRG) in accordance with the present
invention performs the function of combining voice and data into
call records. It performs this function at or near real time. The
CRG, when combined with the metadata normalization module CTI
Server, makes up a system that can be used in current and future
communication recording products.
The CRG is responsible for collecting data from different sources
with respect to portions of a call on various recording input
channels, and merging them together into a unified call record. One
of these sources is the recorder that creates the files containing
media. Another sources provides metadata describing the when, who,
why and where information of a call. This call record metadata
comprises the start and stop times of a segment within a call, as
well as CTI data such as telephone numbers and agent IDs. These
metadata sources include but are not limited to Telephony switches
and Trunked Radio servers. The CRG depends upon the CTI Server to
normalize data from these sources.
FIG. 1 illustrates the relationship between the CRG and the rest of
the system. Since call records are an essential part of the
recording system, there is one CRG dedicated to each recorder and
physically located in the same Voice Server. If other system
components become inoperable, call record generation will remain
functional (albeit at a reduced level).
The CTI server supplies switch events to the appropriate recorder
indicating either the status of calls or providing data for
population. The CTI server provides, along with call record data,
the association between the recorder location (i.e., Voice Server
and recording input channel number) and the switch connection
point. The switch connection point is described as either the
extension for extension side recording or the Trunk ID/virtual
channel (TDM time slot) for trunk side recording. In addition to
this mapping, an agent identification will be supplied for agents
currently associated with this call. The recorder location, switch
identification and corresponding agent are stored in the call
record. The CRG is designed to work with many different
configurations of the disclosed system. These configurations
include: systems without CTI Servers; systems with Real-time CTI
Servers; systems with non-Real-time CTI Servers; recorders with
analog inputs; recorders with digital inputs; recording on the
trunk side of the telephony switch; and any combination of CTI
Servers, Recorder inputs, and recorder positions mentioned
above.
Due to the non-standard operation of telephony switches and
flexibility requirements of the recording device, the CRG must
handle event data arriving in different chronological order. In
accordance with a preferred embodiment, it accomplishes this by
requiring all events to indicate time of occurrence and maintaining
a history of them. A call record can be created solely from either
event sources but when both are present, call records are generated
using recorder information together with CTI data.
It is clear that the use of different data sources and
non-synchronous messages, as required to support various
alternative configurations of the overall system, add considerable
complexity to the CRG. For example, with the many different objects
supplying information for a particular call, the messages from each
can be received in any order. The CRG must be able to accommodate
this requirement. In some configurations, objects supply redundant
information to the CRG. The CRG provides a mechanism for selecting
which information will populate the call record.
In the most basic mode of operation, the CRG has no CTI input and
is recording solely on VOX events from the recorder controller (the
term "recorder controller" is used interchangeably herein with
"Audio Recorder"; both terms refer to the software that primarily
directs the processing of the audio data). VOX is Dialogic
Corporation's digital encoding format for audio samples. This term
is also sometimes used to refer voice-activated initiation of
recording, a process that conserves storage space since a
continuous recording process would include periods of silence.
These VOX events mark the beginning of energy activity on a phone
line and are terminated by the lack of activity. With this
approach, an actual phone call may include several call records. To
address this problem, the recorder waits a configurable holdover
period while silence is present before terminating an active VOX
clip (the term "Recorder" is used interchangeably herein with the
term "Voice Server"; both terms refer to the physical recording
server). The goal is to concatenate parts of a phone call where
gaps of silence exist. The solution lies in determining an
appropriate holdover time so as to avoid merging audio from the
next phone call if it occurs close to the end of the last call.
The next level of operation is where the recorder hardware can
detect telephony signaling such as off hook and on hook. The CRG
has no CTI input from the switch and is recording solely on events
from the recorder controller, but these events mark the beginning
and end of a phone call (off hook and on hook). The resultant call
record reflects a phone call in entirety but lacks much descriptive
data that accompanies switch data.
The highest level of operation involves the use of a CTI Server. In
this configuration, the CRG receives recorder events as well as CTI
events. Since CTI events give the CRG a description of the entire
phone call, information obtained from them drive the creation of
call records. Recorder data describing audio events are absorbed
into the CTI call record whenever audio and CTI times overlap. With
CTI events driving call record generation, non-audio based call
records can be created.
Mixing of recorder and CTI data occurs by comparing ranges of time
indicated. For example, a person whose telephone extension is being
recorded is involved in a phone call for a given period of time.
The recorder events indicating that audio was recording on the same
extension during the same time period are associated with the CTI
metadata for that phone call. Since the data from the CTI Server
may arrive before or after the corresponding recorder events, the
CRG maintains an independent history for each type of data.
For the case where CTI events arrive before the recorder events,
the CTI events are added to the CTI history list. When the
corresponding recorder events arrive, the CTI history list is swept
for matching time ranges and associations are made when they occur.
For the case where recorder events arrive before the CTI events,
the recorder events are added to the recorder history list. When
the corresponding CTI events arrive, the recorder history list is
swept for matching time ranges and associations are made when they
occur.
Previous recording systems stored voice data and metadata in
separate locations. A significant disadvantage to this approach is
that it is left up to the other software subsystems to combine the
information when required. This approach makes the work of other
system features, such as playback and archiving to offline storage,
more complicated and prone to error. By performing this "early
binding" of the audio and CTI data in accordance with the present
invention, such problems are avoided and the above desirable
features are therefore much simpler to implement in a correct,
robust fashion.
When attempting to playback media for a given call record, the
playback mechanism must figure out where the audio for the call
record exists and when determined, retrieve and locate the start
time inside this media. The CRG places this media metadata in
related tables, thus informing the playback mechanism what files
are associated, their location, and what time ranges inside the
file are available for playback.
Most communication systems require an archive mechanism to store
large amounts of data-that cannot be kept online due to capacity
limitations. The CRG used in accordance with this invention assists
with archiving by allowing both call record metadata and the media
files to be stored on the same offline media. Current versions of
recording systems store call record metadata and media files on
separate offline media making restore operations more
complicated.
For enhanced security purposes in a preferred embodiment, the CRG
accesses media files associated with a call record through the use
of media segmentation. A media segment includes, in addition to a
media filename and location, a start time and duration inside the
media file. Media segmentation is necessary when creating CTI based
call records since a call record may involve many recording
locations throughout the life of the call. The specified time range
isolates a portion of the media file that can be accessed through
this call record. This feature is very important when there are
many call records located in one media file. A user attempting to
play back media of a call record, to which he has the permission
for access, may or may not have permission to play back other call
records sharing the same physical file.
The Call Record Generator is responsible for merging CTI search
data and a multitude of voice recording segments together into a
single manageable unit of data. This software includes a flexible
receiver algorithm to allow voice and search data to arrive in
either order, without requiring one to precede the other. Once
combined, the call record can be managed as a single entity, which
greatly simplifies and reduces the work necessary to perform
search, retrieval, and archival operations. This approach also
offers a more natural and flexible framework for controlling
security access to the recordings, on an individual call basis (or
even on selected portions within a call).
As shown in FIG. 15, a recording unit operating with only voice
signaling to guide the creation of its call records could make a
number of fragmented audio segments. When the recording unit is
supplied with CTI search data giving a complete history of the
call's lifetime, and when it is designed to merge the CTI search
data and audio segments into a combined unit of Voicedata.TM., the
results can simplify and reduce the work necessary for a user to
obtain a desired call from the system. Several audio segments can
be grouped together, and can be understood by the system as being
part of the same logical telephone call. It is also possible that a
single audio segment was recorded, even though parts belong to
separate telephone calls, because the delay between stopping the
first call and starting the second call was very brief. Without a
sufficient silence gap, it may appear to the voice recording unit
that this was a continuous segment of audio, rather than belonging
to two separate calls. When the CTI search data is merged with the
audio segments, the system can use this information to recognize
when an audio segment should be split and divided between two
logically distinct calls.
The purpose of the Call Record Generator (CRG) is to collect
information describing multimedia data and store it in a central
location. The CRG produces Master Call Records MCRs) that
encapsulate information describing a phone call as well as the
location multimedia that is associated with it. This description
data comes from a multitude of sources including but not limited to
a Voice Server and CTI Server. Likewise, the design of the system
envisions that there will be a number of possible input sources for
audio recording.
Whatever the means for collecting CTI information, it is
communicated to the rest of the system in a common, normalized
format. The CTI information is passed from the translation modules
to a message router. From that point, copies of the information are
sent to the scheduling and control services and to the CRG for the
appropriate recorder(s). The scheduling and control services are
responsible for starting and stopping the audio recorder, according
to pre-defined rules that are dependent upon time and CTI
information. The CRG is responsible for merging the audio recording
with the CTI information to determine the temporal boundaries of
the call and prepare the Voicedata for storage.
The user workstation typically searches and retrieves records from
the Voicedata storage, and then obtains audio for playback directly
from each recorder's private storage area. The user workstation can
also be used to monitor "live" conversations by communicating
directly with the recorder. The user workstation can also control
the audio recorder indirectly by manipulating the rules used by the
scheduling and control services.
In the preferred embodiment, the user workstation has software that
is configured to display a graphical user interface (GUI) such as
that shown in FIG. 16. The GUI in FIG. 16 uses the information
compiled in the Master Call Record to generate a graphical
representation 1610 of the call, as well as displaying the call
record information in alphanumeric form in a table 1620. Further,
when the call is played back, the displayed segments in the
graphical representation are highlighted to indicate the portion of
the call being played back. For example, in FIG. 16, if the entire
call is played back, when the portion of the call that occurred
between 6:20:08 AM and 6:55:31 AM is played back the bars 1632,
1634, and 1636 are highlighted from left to right as the call is
played back. Thus, as the part of the call that occurred at 6:55:31
AM is reached, bar 1634 is fully highlighted, and bars 1632 and
1636 are highlighted starting from the left and extending to those
points on bars 1632 and 1636 that are directly above and below the
right-hand endpoint of bar 1634. After the played back call reaches
the part that occurred at 6:55:31 AM, the bar 1638 begins to be
highlighted starting at the left endpoint. When the part of the
call that occurred at 7:10:22 AM is reached, the bar 1636 is fully
highlighted. At that point, the bars 1632 and 1636 are highlighted
from their left-hand endpoints and extending to points directly
above the right-hand endpoint of bar 1638. The process continues as
long as the call is being played back, until bars 1632, 1634, 1636,
1638, 1642, and 1644 are completely highlighted.
In alternate embodiments of the subject invention, playback of a
potion of a call can be activated directly from the graphical view
by mouse-clicking or by selecting from a pop-up menu; circular
"pie-charts" show the percentage of time for each party involved
during the lifetime of the call; an animated vertical line scrolls
along to indicate the progression of time when the call whose graph
is being displayed is played back; and miniature pictorial icons
are shown within the graphs to indicate start stop reasons, type of
participant, etc. All of these embodiments are enabled by the data
contained in the Master Call Record.
As a method of managing complexity, the preferred embodiment of the
system uses data abstraction to isolate the internal details of
certain structures to those components which need to operate
directly upon them. Information is organized by the collectors (or
producers) of that data, into a digested form that is more easily
usable by the applications which need to retrieve and process the
data.
For example, the CTI translation modules supply normalized records
to the rest of the system in a common shared format, rather than
exposing the details of various different CTI links. The system
data model is call-centric, containing a detailed cumulative
("cradle to grave") history, rather than event-centric, which would
place the burden of work on the receiving applications. Likewise,
agent information is session-oriented rather than
event-oriented.
Whether collecting information from a CTI link, or recording audio
from a telephone call, a fundamental design advantage for the
system of the preferred embodiment that it operates virtually
invisibly, from the end-user's perspective. The system architecture
is designed to avoid any interference with the normal operation of
a call center environment.
For example, the CTI translation modules are focused exclusively on
collecting and normalizing information that is to be supplied to
the rest of the system. Liability recording systems, and quality
monitoring systems that use "service observance" techniques, do not
require any active call control on the CTI links. Only the
technique known as "dynamic channel allocation" requires active
call control through CTI links to establish a "conference" or
"bridge" session between the audio recorder and the telephone call
participants. When active control is required to implement such a
feature, it can be implemented through a new logically separate
task, without significantly affecting the rest of the system
design. For customers that have existing CTI infrastructure and
applications, the system will not interfere with their existing
operations.
The CRG is responsible for collecting data from the CTI Server,
creating CTI-based call records, and attempting to match those
records with existing recorded audio data. If the CRG receives CTI
information indicating that audio data for the same call resides on
two or more recorders (for example, due to a transfer), records
will be generated for each portion with a common call record ID.
This ID can later be used to query for all of the pieces
("segments") comprising the complete call. Each segment will
identify the recorder that contains that piece of the call.
During playback, a player module connects to a program located on a
Voice Server called the Playback server ("PBServer"). The machine
name of the particular Voice Server which holds an audio segment is
stored by the CRG in the call record table within the Voicedata
storage, and is passed into the player module after being extracted
by a User Workstation's sub-component known as the call record
browser. A call record playback request is then submitted, which
causes the PBServer to query for a specific call record's audio
files located on that physical machine, open them, and prepare to
stream the audio upon buffer requests back to the client software
(the player module) on the User Workstation. If successful, a
series of requests is then issued from the client, each of which
will obtain just enough audio to play to a waveOut device while
maintaining a safety net of extra audio in case of network delays.
Upon a request to "move" within the scope of a call record, the
PBServer repositions its lead pointer to the desired location and
then begins passing buffers from that point. This series of Request
and Move commands continues until the user chooses to end the
session by shutting down the client-side audio player.
As used herein, the term "Call Control" refers to the part of the
metadata concerning the creation and termination of call records.
The term "Media" refers to the actual data that is being recorded.
This term is used interchangeably with audio since the primary
design of the CRG is to support audio recording. However, the CRG
could apply to any data being recorded including multimedia or
screen image data. The term "Metadata" refers to informational data
associated with multimedia data describing its contents. The term
"Call Participant" refers to an entity that is involved in a phone
call. There are at least two participants involved in a call;
namely the calling and called parties. Participants can consist of
people, VRUs, or placeholders for parties being placed on hold. The
term "Recorder Participant" refers to a participant in the MCRs
Participant list who is located at the same connection point on the
Switch to which the recorder input channel is connected. In
accordance with the present invention, there can be more than one
Recorder Participant associated with a call record since
participants can enter and leave many times in a call. For any
given recorder channel, there can only be one matching Recorder
Participant active (not disconnected) at any given time across all
call records associated with that channel. A "VOX-based Master Call
Record contains information contributed by events from the Recorder
alone, in the absence of data from a CTI Server. A VRU is a Voice
Response Unit: an automated system that prompts calling parties for
information and forwards them to the appropriate handler.
Once a recorder channel becomes involved in a phone call, it will
be associated with all subsequent CTI events pertaining to the same
call. This occurs even if the recorder location is no longer
involved in the call. As an example, consider a phone call
involving a transfer. FIG. 16A shows the subject system containing
a CTI Server 710 and Recorder 1640. A recorder channel 01650 is
attached to the extension side to extension 00011622. A phone call
is initiated from the outside by some agent "A" 1602 and initially
connects to agent "B" 1608 at extension 00011622. Agent "B" 1608
places "A" 1602 on hold and transfers him to Agent "C" 1612 at
extension 00021630. The CRG recording extension 00011622 would
receive all update messages with regard to this call since he/she
participated in the call. Descriptive information from the CTI
Server 710 would look like that in table 1600 in FIG. 16B. Audio
clips recorded while agent "B" 1608 was involved in the call are
recorded in a VOX based call record as shown in FIG. 17. The three
media files created from the conversation may overlap with the
recorder participant (agent "B"). At some point, determined by the
order by which recorder and CTI events are received, audio data
information from the VOX call record is absorbed into the CTI MCR
for the times the recorder participant is involved (see the results
after the sweep of the VOX and CTI history lists). For this call
record, audio recorded between times t.sub.1 and t.sub.4 is
absorbed. Any remaining audio is left in the VOX MCR for possible
absorption in other CTI MCRs adjacent in time to this one. Since
extension 0001 in this call record is different from the other
participants in that it is associated with the same switch point as
the recorder channel, he/she is referred to as the Recorder
Participant. From time t4 and on when the Record Participant is no
longer involved in the call, CTI events are still received for that
channel. This allows the system to supply information about the
entire phone call involving extension 0001 that may be of interest
to the customer.
Since the CRG must be prepared to handle messages from different
components arriving in any order, it is designed to collect
information in separate structures. Depending upon the operating
mode of the CRG channel, call records are created from information
collected in one or more of these repositories. The name given for
these structures is Master Call Record (MCR).
The major components of the preferred embodiment contributing
information for call records are the Recorder and the CTI Server.
In alternate embodiments of the subject invention, other multimedia
or screen image data may be provided to the CRG in order to be
merged with descriptive metadata.
Recorder events are assembled into VOX MCRs identified by a unique
sequence number. Individual events contain a sequence number
identifying a specific structure to update (or create). For
example, a recorder event would be used to indicate the beginning
of a new audio segment. While that segment is active, other
messages containing the same sequence number are used to add
metadata to the audio segment. These update events include, without
limitation: DTMF digit data; agent association information; change
of audio filenames holding future audio data; selective record
control; and ANI, ALI, DNIS information. DTMF is Dual Tone
Multi-Frequency and refers to sounds emitted when a key is pressed
on a telephone's touch-tone keypad; ALI is Automatic Location
Identification, a signaling method that identifies the physical
street address of the calling party and typically used to support
Emergency 911 response. Finally, a disconnect message identifies
the end of an audio segment.
Events received from the CTI Server are accumulated in CTI MCRs.
Each event received from the CTI server contains a unique
identifier. Events containing the same unique identifier are
associated with the same CTI MCR. If any VOX MCR contains audio
data that overlaps in time with Recorder Participants in a CTI MCR,
then that audio data is transferred to the CTI MCR. If the
absorption process causes all audio metadata for a VOX MCR to be
consumed, the VOX MCR is deleted from the VOX list. Therefore, call
records generated on the same channel will never have overlapping
audio data. VOX MCRs containing leftover audio not absorbed by CTI
MCRs are either be saved into the central database if of
significant duration or discarded.
Data from a Master Call Record alone is processed into call
record(s) that populate the system's central database. Thus, if the
recorder channel is set up for VOX based recording only or if the
CTI Server is down, VOX MCRs drive call record creation in the
system. Otherwise, the CTI MCRs drive call record creation in the
system.
The VOX and CTI MCR structures are maintained in two separate lists
for each recording input channel. These are the VOX History List
and CTI History List respectively. These lists represent a history
of call activity sorted chronologically. The depth of the history
list is driven by a configurable time parameter indicating the
amount of history that should be maintained. By maintaining a
history, the CRG tolerates events received in any order as long as
received within the time boundaries of the history list. Some CTI
Servers obtain data from SMDR type switches which report entire
phone calls at the end of the call with a summary message.
Maintaining a history buffer for VOX MCRs allows us to hold onto
audio data for a period of time to allow later CTI summary messages
to consume (absorb) the associated audio.
The MCR has status fields associated with them indicating its
current state. At an installation involving real time CTI events,
when a recording input channel receives a CTI event, it may
indicate that a participant connected at the same telephony switch
location as the recorder (Recorder Participant) is active in the
call. The MCR is considered active as long as there is a Recorder
Participant still active in the call. During this period, any new
audio arriving on this channel is associated with the MCR. When a
Recorder Participant leaves the call, the MCR becomes inactive.
Since any Recorder Participant can become involved in the
conversation at any given time through transfers or conferences,
the MCR can transition into and out of active state many times
throughout the phone call.
Another field in the MCR indicates the overall status of the call.
This flag, called m_bComplete, indicates when the phone call is
over. An MCR is considered incomplete as long as there is at least
one participant still active in the call. When there are no
participants active in a MCR it is considered to be complete.
Therefore, calls created in real-time will start as incomplete and
at some point transition into completed state. When an MCR enters
complete state, a Closed Time variable is set to the current time.
This time is used in maintenance of the History List. A closed MCR
is allowed to stay in the History list for a configurable amount of
time before it is deleted. During this window of time, events
arriving out of timely order are allowed to update the MCR. Once
this configurable amount of time expires, the MCR is updated in the
local database, marked complete, and deleted from the History
List.
When the CRG starts, it initializes, for each recording input
channel, a location which identifies where it is attached to the
telephony switch. Each recorder location contains status fields
describing the state of the switch and CTI server involved. These
fields are m_SwitchStatus and m_MetadataServerStatus respectively
and are set to "down" state until an event is received that
indicates otherwise. When a message is received indicating a change
of state, all associated recorder locations are updated with the
new state value. Any changes in operation are processed upon
receipt of the next event for the channel.
Another configuration setting indicates what type of external
sources are allowed to populate call records created on a record
channel. This setting, m_ExternMetaDataSource, is set to zero when
a record channel is to be driven by recorder events only. It is set
to non-zero when external events are allowed to generate MCRs.
The CRG is able to react to a variety of situations that may arise.
For example, when the CRG first initializes and a record channel is
configured to receive CTI input, how are call records generated if
the CTI server is not running? What if the CTI Server is running
but the communication path to the recorder is down? The CRG must
also be able to react to external parts of the system, that it
normally relies on for input, being temporarily unavailable for
periods of time. In accordance with a preferred embodiment, the CRG
handles these situations by operating in different modes: Initial,
Degraded, and Normal. These modes are applied individually to each
channel in the recorder.
Initial Mode: When a recorder starts up, there can be a
considerable amount of time before the rest of the system becomes
operational. The CRG must be ready to handle events coming from the
Recorder immediately after startup. Therefore, the CRG must be
ready to accept recorder metadata without supportive information
from the CTI server. VOX MCRs are created from these recorder
events and are stored in the VOX History List. When VOX MCRs are
completed, they are made persistent in the Local Data Store.
The CRG system will remain in this mode until all of the following
conditions occur: (1) the CTI server becomes available; (2) the
switch being recorded by this channel becomes available; and (3) a
configuration option for the channel indicates it is to be driven
from an online CTI server and switch.
Degraded Mode: If a record channel is configured to be driven from
a CTI source, only CTI MCRs are entered into the database. These
CTI MCRs absorb any recorder metadata that intersects with the time
ranges of the CTI events. No VOX MCRs are made persistent. If,
however, the CRG detects that the CTI Server, switch, or associated
communication paths are down, the channel enters Degraded mode.
This mode is similar to Initial mode in that VOX MCRs are made
persistent when completed. Any CTI MCRs that were left open at the
time the CTI Server went down are closed and updated for the last
time. The recorder channel will remain in this state until the
three conditions indicated in "Initial Mode" are met. Only then
will the recorder channel transition into Normal mode.
Normal Mode: Under normal operating procedures in a system with a
CTI server and switch online, MCRs are created whenever a VOX or
CTI connect event is received and stored in the appropriate list.
For each VOX message received, the CTI History List is swept to see
if audio metadata can be absorbed by a matching MCR. Any remaining
audio data is placed in a VOX MCR. For CTI events involving updates
to Recorder Participants, the list of VOX MCRs is swept to see if
audio metadata can be absorbed. CTI MCRs are made persistent to the
Local Datastore when first created, upon significant update events,
and when completed. VOX MCRs are not made persistent to the Local
Datastore as they should be completely absorbed by CTI MCRs. There
is a configuration parameter that can enable leftover VOX MCRs to
be made persistent when they are removed from the VOX MCR history
list.
Transitions from Initial/Degraded to Normal Mode: When a CRG
channel is in Initial or Degraded mode, VOX MCRs are recorded into
the Local Data Store when completed. If notification is received
indicating a recorder channel meets the three criteria indicated in
"Initial Mode", the channel is set to Normal mode. From this point
on, only CTI based MCRs are made persistent and VOX MCRs will be
absorbed by the VOX events. Since CTI events represent an
accumulated history of a phone call, prior events occurring while
the connection between the CRG and CTI Server was lost (or was not
yet established) are nonetheless summarized in each update message.
The time spans of Recorder Participant(s) are compared to audio
data in the VOX MCR list, with any overlaps causing the audio data
to be absorbed. In this way, any audio data that occurred while a
connection to an external component is temporarily unavailable will
still be capable of being correctly associated.
Transitions from Normal to Initial/Degraded Mode: When the CTI
server and switch becomes available for driving the call record
creation and processing, the CRG channel enters into Normal mode. A
heartbeat message is used to periodically update the status of the
switch and CTI Server. When the heartbeat is lost or there is a
message indicating one the components has gone down, the recorder
channel switches to Degraded mode. The CRG will still create and
maintain MCRs in the VOX list and force MCR closure on open CTI
MCRs as they pass out of the CTI history buffer. The sweeping
action of audio metadata among incomplete CTI MCRs will cease,
preventing all future audio data from being absorbed by it. VOX
MCRs are made persistent in the database when they leave the
history buffer.
Trunked Radio Mode: In an alternate embodiment of the subject
invention, fields in the call record structure are added to support
trunking radio. Information contributing to these fields may be
obtained from communications with a Motorola SmartZone system. This
system uses the Air Traffic Information Access (ATIA) protocol to
communicate metadata related to radio activity. The embodiment has
a trunking radio server similar to the CTI server that provides an
interface between the SmartZone system and the recorders of the
preferred system. This server provides the normalization of data
and distribution to the correct recorder. There are currently two
modes of operation of the Motorola trunking radio system that are
discussed below.
Message Trunking: In this mode, when a radio is keyed, it is
assigned a particular frequency to communicate on. When the radio
is de-keyed, a message timeout timer (2-6 seconds) is started. If
another radio in the talk group keys up during this time, the
controller uses the same frequency for transmission and resets the
timer. The conversation will remain on this frequency until the
timer is allowed to expire. During this time, all events that are
reported with respect to this conversation will have the same call
number associated with them. Therefore, the concept of CTI based
call records with many participants has been applied to Message
Trunking.
If the timer is allowed to expire, future radio transmissions will
be assigned to another frequency and call number. The server needs
to detect this occurrence and properly terminate a call record.
Transmission Trunking: Transmission Trunking does not use the
holdover timer mechanism used in Message Trunking. When a radio is
keyed, it is assigned a particular frequency for transmission. When
de-keyed, the channel frequency is immediately freed up for use by
another talk group. Therefore, a conversation can take place over
many channels without a call number to associate them. The concept
of VOX based call records which contain one radio clip per MCR is
used in this mode.
Selective Record: There may be certain phone calls involving
extension or agents that are not to be recorded. Selective Record
is a feature that tells the system to refrain from recording a call
while a certain condition exists.
Virtual CRG: MCRs can exist in the subject system's database that
have no audio associated with them. These non-audio MCRs can be
created due to different features of the subject system. Some
customers may require that all CTI data coming from their switch be
saved even though they are not recording all extensions or trunk
lines. By creating records from the CTI data alone, in the absence
of recorded audio, this mode of operation can provide the customer
with useful information for statistical analysis or charting
purposes. Likewise, records created based upon CTI data alone may
provide a useful audit trail to verify the occurrence of certain
telephone calls, analyze traffic patterns, or to perform other
types of "data mining" operations. In that case, a CRG is
associated with the CTI Server mechanism to receive all CTI events
that are not matched to a specific recorder. These CTI MCRs are
made persistent to the Central Database upon call completion.
Call Record Structure: Call record start and stop events originate
from two independent sources: the Recorder and the CTI server. The
CRG must perform some method of merging events from these two
sources in such a way that the resultant call record contains the
best information available. CTI server events are advantageous in
that they provide more information than the recorder and can also
accurately determine a call record boundary. Recorder based events
are a subset of CTI server events and can only distinguish call
record boundaries based upon VOX or off/on hook. The recorder has
advantages in that since it is in the same box as the CRG, receipt
of these events is guaranteed as long as the recorder is running.
The main purpose of the assembly process is to leverage the
information coming from the CTI server in such a way that the
entire phone call is assembled into one Master Call Record (MCR).
The structuring of call records is weighed towards trunk side
recording with the services of the CTI server driving call record
creation. This type of configuration enables the system to
summarize phone calls in the most effective manner. The manner in
which the structure of the MCR designed to achieve this goal is
discussed below.
Master Call Record: The MCR holds information accumulated for all
events received necessary for archiving to the local data store. It
consists of individual fields that are global to the entire call
record as well as lists of specific information. Global information
includes identifiers for the call record, the start and stop times
of the entire call, the recorder location with respect to the
switch, and flags indicating the call record status.
Lists included with each MCR contain the following information:
Media File List--List of media filenames that make up the call
(e.g., telephone or radio communications); Screen Data Capture File
List--List of screen image files associated with audio on this
channel; and Participant List--List of participants involved in
this call.
The MCR is populated from events received from the CTI Server and
Recorders. The following table shows the fields in the MCR, in a
preferred embodiment, their data types, description and if they are
stored in the database.
Master Call Record structure. Type(max Name length) Archive
Description m_CallRecID string.dagger. Y Unique ID (UUID)
pertaining to entire call (Ctl and Trunk Radio server provides same
ID for call parts that are related to the same conversation.)
m_MetaDataSource BYTE Y Indicates the source used to populate call
record information. 0 = none 1 = CTI 2 = Trunking Radio
m_bCallComplete bool N Indicates the end of a call. (i.e., there
are no more active participants involved) m_bCall- bool N If true,
MCR has been in a complete HoldoverExceeded state for a time period
exceeding the configured Call Holdover time. m_bMetadataHoldoverEx
bool N If true, MCR has been inactive for a ceeded time period
exceeding the configured Metadata holdover period. Used to allow
completion of MCRs that haven't been updated for long periods of
time possibly because of missed events. m_bLastUpdate bool N true
when the CRG has decided to send the last update of this MCR. Used
to prevent any future updates. m_bDontArchive bool N Indicates
whether this call record is to be archived by data store. Certain
record features such as selective record may prevent us from
storing this call record. m_CallDirectrion BYTE Y Indicates call
origin Outbound = 0x12, Inbound = 0x21, Internal = 0x11, Unknown =
0x44 m_CustomerNumber string.dagger. Y Variable length character
array dedicated to information the switch may provide with the
call. For custom call record support. (e.g., account number)
m_pRecLoc RecorderLocati N Pointer to recorder location descriptor
on associated with this channel. (see RecorderLocation class)
m_SSFile list.dagger. Y List of TimestampedEilename (see below)
objects representing Screen Data Capture filename(s) associated
with a call record m_Participants list.dagger. Y Array of
CallParticipants (see below) describing all participants involved
in the call m_XactionSema HANDLE N Semaphore used to lock this MCR
from being modified by any other threads. m_SemaTimeoutVal unsigned
long N Maximum time thread is blocked on m_XactionSema access
before returning. m_bModified bool N Set whenever MCR is changed in
a way that requires update to the Local Data Store. VOX Call Record
(Derived) m_dwVoxCrNum DWORD N Sequence number of first VOX MCR
associated with this OTI MCR (if applicable). m_bVoxInProgress bool
N Indicates this VOX clip is still active (i.e., End time is
default time.) m_CreationTime time_and_date N Holds time at which
the call record was .dagger. created. Used for debugging purposes
to measure how long a call record is alive. m_CloseTime
time_and_date N Local time at which MCR was marked .dagger.
complete. Used for determining when call record is ready for
archive. m_MediaFiles list N List of TimestampedFilename classes
representing multimedia files used to store data with respect to
this call record. m_CtiInfo CtiInfo N Class containing CTI type
data associated with call record. Base Call Record (Derived)
m_wVersion WORD N Version number of call record. m_StartTime
time_and_date Y Start time of call record .dagger. m_EndTime
time_and_date Y End time of call record .dagger. RecorderLocation
m_MetadataServerStatus BYTE Indicates the status of the metadata
server driving call records for this particular recorder location.
This source is in most cases the CTI server but can be other
servers such as Trunking Radio Server 0 = "down", 1 = "up"
m_SwitchStatus BYTE Indicates the status of the telephone switch
providing call record information for this particular Recorder
Location. 0 = "down", 1 = "up" m_ExternMetaDataSource BYTE
Indicates what external source (if any is contributing call record
meta data for this channel 0 = None (recorder only) 2 = CTI server
m_ChanID ChannelIdentifier Class identifying recorder channel.
m_SwitchID Switch Identifier Class identifying switch connection
point. m_SwitchChars SwitchCharacteristics Class identifying
characteristics of switch needed by CRG. SwitchIdentifier
m_SwitchNum WORD Number identifying switch m_wTrunkID WORD
Identification of trunk line attached to switch. (Valid only if not
equal to -1) m_dwVirtualChannel DWORD Identifies time slot of
digital line (T1 or E1) of interest. (Valid only if TrunkID is not
equal to -1) m_Extension string.dagger. (6) Extension number (Valid
only if m_wTrunkID equals -1) ChannelIdentifier m_wNode WORD Unique
number used to distinguish between multiple Voice Servers.
m_wChannel WORD Unique number used to distinguish between multiple
recording input channels within a Voice Server. m_bSignalSupport
bool Indicates if hardware associated with this channel supports
on/off hook signaling. SwitchCharacteristics m_bTimeSynced bool
Indicates if switch is synchronized with the system. m_bRealTime
bool Indicates if switch provides CTI info in realtime (true) or
batched and sent periodically (false) m_iCmdTimeOffset int Value
that indicates any known time offset between events received at the
switch versus the time the similar signal is received at the
recorder. This value will be used to adjust CTI generated
timestamps before comparing to recorder events m_iSwitchTimeOffset
int For switches that are not time sync'd with the system, this
value indicates any known time offset between the switch and the
system time. This can be utilized if has some way of updating the
time delta between switches and our system on a periodic basis.
CtiInfo RingLength WORD Time (in sec) between first ring signal and
off hook. DTMFCode string.dagger. (50) DTMF codes entered during
conversation Name Type Description TimeStampedFilename
m_AFStartTime Time_and_date.dagger. Start time of audio file
m_StartTime Time_and_date.dagger. Start time of interest m_EndTime
Time_and_date.dagger. End time of interest m_SegStartTime
Time_and_date.dagger. Start time of segment inside file absorbed by
this MCR. m_SegEndTime Time_and_date.dagger. End time of segment
inside file absorbed by this MCR. m_PathName string.dagger. (36)
Path describing the Voice Server and directory location where the
audio files are located. m_File Name string.dagger. (36) GUID-based
name that uniquely identifies a specific audio segment's recording
file. m_wFileType WORD bitmap indicating types of
media associated with MCR. bit Data 0 - Audio Present 2 - FAX
Present 3 - Video Present 3 - Screen Capture data Present
m_wFileFormat WORD Recording format of media data, as defined by
Microsoft Corporation's multimedia description file "mmreg.h"
m_bNew bool Used by local data store to indicate whether this
record should be inserted (true) or updated (false) into the
database. m_bDiscard bool If true, don't allow playback or
archiving of this media. Used for the Selective Record feature.
m_dwVoxCrNum DWORD Sequence number corresponding to VOX call record
that provided this media. m_iAssocPart int Index of Recorder
Participant in the Participant list causing this media file to be
associated with this MCR. CallParticipant m_AgentID string.dagger.
(24) Registered ID of agent at extension (CTI) or Radio Alias
(Trunking Radio). m_Number string.dagger. (24) Full telephone
number of the participant (i.e., ANI, DNIS) m_Console
string.dagger. (10) Seating position of participant that can
consist of one or more stations (CTI) or Talkgroup ID (Trunking
Radio). m_Station string.dagger. (10) Unique telephone set.
Possibly with multiple extensions m_LocRef BYTE Describes the
location of participant with respect to the switch. (1 = internal,
2 = external, 3 = unknown) m_SwitchLoc Switchldentifier Class
identifying the position of a participant relative to the telephone
switch. m_StartTime Time_and_date.dagger. Time participant joined
the call m_EndTime Time_and_date.dagger. Time participant left the
call m_ConnectReason BYTE How participant joined the call
NotConnected = 0, NormalStart = 1, ConferenceAdd = 2, TransferRecv
= 3, UnknownConnect = 9 m_DisconnectReason BYTE How participant
left the call NotDisconnected = 0, NormalEnd = 1, ConferenceDrop =
2, TransferAway = 3, OtherPartyHangup = 4, UnknownConnect = 9
Changed Bool Indicates if recent change in CTI message. (not
archived) Trunking Radio Only Information SourceSiteID BYTE Site
number that is currently sourcing audio on active call. ZoneID BYTE
Zone at which participant is currently located. CIUNumber BYTE
Console Interface Unit. Translates l2kbit into clear audio &
vice versa. CDLNumber BYTE Channel associated with CIU DIUNumber
BYTE Digital Interface Unit. Translates ASTRO clear secure data
into analog audio & vice versa. DBLNumber BYTE Channel
associated with DIU .dagger.Objectspace data types
Unused string fields are null. Unused number fields are set to
zero
The version number is used to indicate the structure of data
contained within the call record.
In order to maintain compatibility with future versions, changes to
call record structures will be performed in an additive nature.
That is, current members of the call record will not change in
position, size, or meaning.
Each call record will contain a list to store participant
information. There will be at least two participants in a call
record; the calling and called parties. Any additional connections
that are conferenced in or transferred to are appended to the end
of this list.
Only one active VOX and CTI based Master Call Record is allowed per
recording input channel at any given time.
CRG Software Architecture
FIG. 18 shows the processing threads and data structures that
comprise the CRG module in a preferred embodiment.
Event Processing: when the CRG is created and initialized, three
threads are created. These threads are the CRG Event Processor
thread 1810, Facade thread (The terms "facade," "facade," and
"fascade" are used interchangeably in this disclosure) 1812 and
Local Data Store thread 1816. Additionally, three message queues
are created and are known as the Recorder 1824, Facade 1832, and
Data Store 1844 queues, respectively. These queues enable the
processing of various input messages in a de-coupled fashion within
the CRG, so that any delay in one area of communications does not
affect the processing of another area. Each thread is described
below.
Event Processor Thread: the Event Processor is the primary thread
of the CRG module. Its responsibilities include reading any
messages placed in the Recorder 1824 and Facade 1832 queues. The
processing activities that occur in response to these messages
cause updates to be made to call records belonging to one of the
recording input channels 1856. If these changes cause a call record
to be completed, a message is sent to the Date Store queue 1844
requesting that the call record be made persistent in the local
database. This thread is also responsible for processing state
change messages, that cause memory resident structures to be
refreshed or to shut down the CRG module.
Facade Thread: The Facade thread handles messages that come from
outside the Voice Server. Its primary function is to look for
messages placed in the CRG's external Microsoft Message Queue
(MSMQ) 1864 where events may arrive from other components within
the overall subject system. Upon receipt of a message, the Facade
thread reads the message, translates it into an appropriate format
for the CRG's internal data structures, and places the translated
copy in the Facade Queue 1832. This thread is known as the Facade,
because it manages the external interactions of the CRG with the
other components within the subject system.
Local Data Store Thread: The Local Data Store thread 1816 processes
requests from the CRG Event Processor thread 1810. The primary
purpose of the Local Data Store thread 1816 is to take internal
Master Call Record (MCR) structures and translate their contents
into structures compatible with database technologies, such as
Microsoft SQL Server, or comparable types of storage means. These
resultant structures are stored within the database in order to
make the call record persistent.
Characteristics of some switches mandate that the CRG be able to
handle CTI events that are not real-time. Some switches batch
events and send them out periodically. CRG configuration settings
that limit the history list by time must be set long enough to
accommodate the switch characteristics. Therefore, call records
that are generated between switch reports (via recorder events)
will not be finalized until a configurable time period (window)
after which the call record terminated. This window
(CallHoldoverPeriod) needs to be set to a minimum of the period of
time between switch reports. Once a call record leaves this time
window, it is marked as read-only and committed to the local data
store.
A situation that must be dealt with is when the telephone switch is
not time synchronized with the rest of the system. To facilitate
the merger of recorder and switch events effectively in
non-time-synchronized systems, alternate embodiments of the subject
system are described.
One alternate embodiment of the subject system has a mechanism that
synchronizes the clocks in the system (manually or automatically)
on a periodic basis. This must guarantee time skews of less than
some small and known quantity. A second embodiment has a mechanism
for measuring the time delta between the switch and the subject
system. This value is updated periodically and used by the CRG
during the merging process. A third embodiment implements a
combination of the first two.
During the call record merging process, a global time delta is used
to adjust switch event time stamps before comparing to existing
call record data.
The following paragraphs define the types of events the CRG is
designed to accept and process. These events may cause the CRG to
initialize, process metadata into call records, or prepare the
system for shutdown.
The Master Controller (a sub-component of the present system's
Scheduling & Control Services) supplies system events. The
Master Controller notifies the CRG of system related changes such
as configuration changes, CTI server status and system shutdown
events. The CRG changes its behavior based upon events received
from the Master Controller.
System Events: The CRG provides an interface that allows the client
application to control its states of operation. This is
accomplished with an interface class that is used by most system
components in the subject system. The interface is named IProcCtrl
and supports the following methods: Initialize( ); Start( ); Stop(
); Pause( ); Resume( ); Ping( ); and Shutdown( ).
In addition to these methods, the CRG supports two event messages
that inform it of status changes that are needed to either update
its memory resident configuration information or change its mode of
operation. These methods are CtiStatus and AgentExtensionStatus.
Each method is described in the following paragraphs.
Initialization Event: This method is the first method that should
be called after the CRG has been created. When the CRG object is
created, it retrieves configuration information from the subject
system's database. This information describes the number of
channels in the recorder, the switch location where each channel is
connected, any fixed associations of telephone extensions or agent
identifiers. Also included are parameters that determine the
behavior of the CRG. Threads are spawned to handle the processing
of CRG events, communicating with external metadata contributors,
and processing information into the Call Records tables. These
threads are created in a suspended state and require the Start or
Resume commands to begin processing activity.
Start Event: This method should be called after the Initialization
event. It resumes all threads of the CRG enabling it to process
incoming events.
Pause Event: This method suspends all threads of the CRG.
Resume Event: This method is called after the Pause command to
enable all CRG threads to continue processing.
Ping Event: This method is used by client applications to test the
connection to the CRG. The method simply returns a positive
acknowledgment to let the client know that the CRG is still
running.
Shutdown Event: This method notifies the CRG when the subject
system is shutting down so that it can cleanly terminate itself.
The shutdown event supports a single parameter (ShutdownMode) that
indicates how it should shutdown.
If the ShutdownMode is specified as "Normal", all pending events
read from the input event queues and processed into the call
records, any open call records remaining are closed at the current
time and written to the database.
If the ShutdownMode is "Immediate", input event queues are cleared
without processing into call records, open call records are closed
and written to the database.
Once these actions are completed, the CRG threads terminate. At
this point, it is now safe for the client application to release
the resources of the CRG.
Stop Event: This method is implemented for consistency with the
common interface of IProcCtrl. The CRG has no purpose for this
method and just returns a positive acknowledgment.
CtiStatus Event: This event informs the CRG of the operational
status of the CTI server that is providing it with telephony
metadata needed for CTI call record generation. The Scheduler
component of the subject system is responsible for maintaining a
heartbeat with the CTI server to detect when connection has been
lost. Any changes in CTI server status result in a CtiStatus
message directed at the CRG.
This message contains one parameter that indicates the new state of
the CTI Server. If the parameter indicates that a CTI Server has
become operational, recording input channels associated with the
CTI Server change from "Degraded" mode of operation of "Normal"
mode. If the parameter indicates that the CTI Server is not
operational, recording input channels associated with the CTI
Server change from "Normal" mode of operation to "Degraded"
mode.
AgentExtensionStatus Event: This event indicates that a change in
one of the Agent or Extension tables has occurred. Since the CRG
uses these tables to associate with recorder channels, the memory
resident version must be updated. Therefore, this event causes the
CRG to read these tables and update its memory resident copy.
Call Record Events: When a call record event is received, the
message is interpreted to determine which recording input channel
may be affected. Any filtering necessary on a per channel basis is
performed at this stage. Call record events are then dispatched to
the appropriate Call Record Channel Manager. There is a separate
call record channel manager, which is a software sub-component of
the CRG, for each recording input channel in a Voice Server. There
are three messages that directly contribute to the creation and
completion of call records. One comes from the CTI Server in the
form of a CTI Event. The other two originate from the recorder and
are the VoxSummary and VoxDisconnect messages. Each message is
described in detail below.
CTI Event: The CTI Event is a message originating from the CTI
Server software module that processes the information received from
the telephone switch. The message details each participant involved
with the phone call as well as information global to the call such
as ring duration and DTMF codes. A CTI event message is sent to the
CRG whenever a change in participant status occurs as well as when
new ones enter the call. The messages are cumulative in that all
information of the previous messages is contained in the new one
with any additions included. This makes for a more robust system in
cases where one of the messages is lost.
The pseudo code for processing a CTI event is shown below:
Pseudo code for CTI Event // --------------- CTI Event (BEGIN)
----------------- // Don't process OTI events if we're not in
correct mode Is this recorder channel configured to receive OTI
event data? {// Yes `Does this event match an MCR in my CTI History
list? {// Yes Update MCR participants with matching one in CTI
event Add any new participants to MCR. UpdateMediaFiles() (see
pseudo code) }// End - Does this event match an MCR in my history
list? Otherwise { Create new MCR Initialize MCR Start time from
Oldest Participant Start time in event Copy participants from event
to message to MCR. // Now that we've updated the participants, see
if // we need to change media file associations. UpdateMediaFiles()
(see pseudo code) Insert new MCR Into Cti MCR history list. } Are
there any participants still active? Mark MCR as active Otherwise
Mark MCR as complete }// End - Is this recorder channel configured
to receive CTI event data? // --------------- CTI Event (END)
--------------- // --------------- UpdateMediaFiles (BEGIN)
--------------- for each Recorder Participant in the MCR { Is this
not a new Recorder Participant? { // This participants start and/or
end time may have been adjusted. // See if audio previously
absorbed by it has to be returned to the VOX history list
FindGiveBackMediaFiles() (see pseudo code below) } for each MCR in
VOX History list with a time range that overlaps with this recorder
participant { for each media file in this VOX MCR that's timespan
overlaps with this recorder participant { CheckAndApplyMediaFile()
(see pseudo code) }End - for each media file in this VOX MCR that's
timespan overlaps with this recorder participant Did we consume all
audio in this VOX MCR? Remove VOX MCR from History list and delete
it. }End - for each MCR in VOX History list that's time overlaps
with this recorder participant }End - for each Recorder
Participants in the MCR GiveBackAudio() (see pseudo code) //
--------------- UpdateMediaFiles (END) --------------- //
--------------- FindGiveBackMediaFiles (BEGIN) --------------- for
each media file associated with the given CTI MCR { Was this media
file contributed from the given recorder participant? {//Yes if
media file lies completely outside recorder participant timespan?
.vertline.<------Participant Timespan
---------------.fwdarw..vertline. .vertline..rarw.---Media File
timespan---.fwdarw..vertline. Move this entire media file to the
Giveback list Otherwise, If media file start time is before the
recorder participants start time?
.vertline..rarw.----------Participant
Timespan----------.fwdarw..vertline.
.vertline..rarw.----------Media File
Timespan----------.fwdarw..vertline. Make a copy of this media file
and set its end time to the participants start time. Add media file
to giveback list. Set original media files start time to that of
recorder participant. Otherwise, if media file end time is after
the recorder participants end time?
.vertline..rarw.------Participant Timespan------.fwdarw..vertline.
.vertline..rarw.------Media File Timespan------.fwdarw..vertline.
Make a copy of this media file set its start time to the
participants end time. Add media file to giveback list. Set
original media files end time to that of recorder participant. }
}End - for each media file associated with the given CTI MCR
//---------------FindGiveBackMediaFiles (END)--------------- //
--------------- GivebackAudio (BEGIN)--------------- // Sweep
through VOX MCRs re-populating any giveback audio for all audio
portions in given back list { if we find the VOX MCR this audio
originally came from? {// Yes Attempt to merge the give back media
file with an existing VOX MCR media file that's start or end time
is adjacent to this ones. Otherwise, associate this media file with
the VOX MCR. }End - if we find the VOX MCR this audio originally
came from? Otherwise { // Original VOX MCR containing this audio
file doesn't exist anymore. Create a new MCR. Associate the
giveback media file with the new MCR Insert MCR into VOX History
list } }End - for all audio portions given back //
---------------GivebackAudio (END)------------------------------ //
---------------CheckAndApplyMediaFile (Begin) Does Recorder
Participant span the entire media file?
.vertline.<--------------- Participant
Timespan--------------------.fwdarw..vertline.
.vertline..rarw.---Media File timespan-.fwdarw..vertline. Move
media file from VOX MCR to Cit MCR. Otherwise, Does Recorder
Participant overlap with media file start time?
.vertline.<---------------Participant
Timespan---------------.fwdarw..vertline.
.vertline..rarw.---------------Media File
timespan---------------.fwdarw..vertline. Make a copy of this media
file and set its end time to the participants end time and
associate with recorder participants MCR. Set the original media
files start time to the recorder participants end time. Otherwise,
Does Recorder Participant overlap with media files end time?
.vertline.<-----Participant
Timespan---------------.fwdarw..vertline.
.vertline..rarw.------------------- Media File
timespan------------------------.fwdarw..vertline. Make a copy of
this media file and set its start time to the participants start
time and Make another copy of this media file and set its start
time to the participants end time and associate with VOX MCR. Set
the original media files end time to the recorder participants
start time. // ---------------CheckAndApplyMediaFile (End)
VOX Summary Event: The VOX Summary Event is a message originating
from the recorder associated with this CRG. It can be used in one
of two ways.
The primary use of this message is to indicate the start of audio
activity in real-time. When used in this mode, the VOXSummary
command indicates the beginning of audio activity. But since the
activity is not complete, the end time is set to indicate that the
VOX segment is incomplete. The end time of incomplete media file is
also set in this way. In this case, a VOX Disconnect message is
required to complete the end times.
The second mode is used to indicate a history of audio activity.
The VOX Summary start and end times reflect the period of time
covered by all accompanying media files. The media files also have
there respective start and end times filled in. This message is
complete and thus requires no follow up messages. The VOXSummary
message is shown below.
Field Name Description VOX Summary Message Format Channel Recorder
channel of audio activity VOXCrNum Sequence number used to
correlate related VOX events. StartTime time at which audio
activity first started EndTime Time at which last audio activity
ended. Media Files list of multimedia filenames used to store data
with respect to this call record. (see below for details)
RingLength Time from start of ring to off hook (in sec) DtmfCodes
String of DTMF codes detected during VOXSummary period
ConnectReason Indication of why VOX segment was started
DisconnectReason Indication of why VOX segment was terminated Media
File Structure FileStartTime Time corresponding to first byte of
audio data in a file. StartTime Time corresponding to first byte of
audio at which activity occurred EndTime Time corresponding to last
byte of audio at which activity occurred FileName String containing
name of audio file. PathName String describing the location of
audio file. iAssocPart Used by the CRG to indicate with which
Recorder Participant this audio segment is associated, when it is
part of a CTI-based MCR. dwVOXCrNum Used by CRG to indicate which
MCR in the VOX History list this audio segment originated.
The pseudo code for processing a VOX Summary event is shown
below.
// ------------------------------- VOXSummary (BEGIN)
------------------------ --------- Is this recorder channel
configured to receive CTI event data? {// Yes // Attempt to merge
media files in message with CTI based MCRs for each CTI MCR in
History list { If any of the given media files fall inside the
timespan of the Cti MCR? { // Yes // Merge media files with
overlapping recorder participants in CTI MCR for each given media
file in VOX Summary message { for each recorder participant in the
given CTI MCR { CheckAndApplyMediaFile ( ) (see psuedo code) } End
- for each recorder participant in the given CTI MCR } End - for
each given media file } End - If any of the given media files fall
inside the timespan of the Cti MCR? } End - for each CTI MCR in
History list Remove media files from VOX Summary message that are
completely consumed } Any unabsorbed audio remaining in message? {
// Yes Create MCR for remainder of audio. Insert MCR into VOX
History List } // --------------------------------- VOXSummary
(END) ------------------------- ------------------------
VOX Disconnect Event: The VOX Disconnect Event is a message
originating from the recorder associated with this CRG. It is used
to terminate a VOX segment that has been started by a real-time
VOXSummary message.
The VOXDisconnect message is shown below.
VOX Disconnect Message Format Field Name Description Channel
Recorder channel of audio activity VOXCrNum Sequence number used to
correlate related VOX events. Time End time of the VOX segment.
Also indicates the end time of open media file. DisconnectReason
Indication of why VOX segment was terminated
The pseudo code for processing a VOX Disconnect event is shown
below.
// ------------------------- VOXDisconnect (BEGIN)
-------------------- ----------------------- Is there a MCR in VOX
History list with the same sequence number? { // Yes // Close and
update all media files in both VOX and // MCR list related to this
one Close Active media file in VOX MCR at given message time
UpdateFromMediaFile ( ) // Update any CtiMCRs that absorbed the
audio file closed. for each MCR in CTI History list { // Attempt to
merge audio with MCR. // Look for matches with audio filenames. for
each media file in Cti MCR contributed by this VOX clip { Close
media file at given message time // Now that we've closed it, does
this media file still // belong with this CtiMCR? Does media file
still fall in time span of MCR? { // Yes CheckAndApplyMediaFile ( )
(see pseudo code) } Otherwise { Remove media file from MCR list and
discard } } End - for each media file in MCR contributed by this
VOX clip } Close VOX MCR and mark as complete } //
------------------------- VOXDisconnect (END)
----------------------- -----------------------
//-------------------------- UpdateFromMediaFile (BEGIN)
------------- ----------------------- // Look for matches with
audio filenames for each media file in MCR contributed by this VOX
clip { Close media file at given message time // Now that we've
closed it, does this media file still // belong with this CtiMCR?
Does media file still fall in timespan of MCR? { // No
CheckAndApplyMediaFile ( ) (see psuedo code) } Otherwise { Remove
media file from MCR list and discard } } End - for each media file
in MCR contributed by this VOX clip // -------------------------
UpdateFromMediaFile (END) ----------------
-----------------------
Data Events: Data events are appended to the currently open
associated call record. For CTI data events, this pertains to a
currently open MCR based upon CTI connect events and containing a
matching call record ID. For VOX data events, the currently open
VOX call record is affected. If an open call record doesn't exist,
an error condition is reported.
Correction Events: Correction events exist to remove a previous
alteration to a call record after it has already been populated.
One reason for such an event is to support selective record. An
audio file that cannot be recorded due to customer or legal reasons
might need to be removed from the call record or the entire call
record might need to be deleted. The VOX event for a filename might
have already been processed into a call record before the selective
record mechanism has determined it not to be recorded.
Selective Record (Exclusion): Selective Record is an important
feature of the subject system, imposed by customer requirements. If
the customer does not want certain participants recorded when they
become involved in a recorded call, the CRG must exclude any audio
associated with the call record for that participants' time of
involvement. Implementing this feature is complicated by the
varying characteristics of customer switches. If the telephone
switch environments report events in real-time, recording of media
can be prevented by turning the recording input channel off during
the selective record participants' time of involvement. However,
what happens when events are not reported in real time from the
switch? The answer lies in the sweeping action of the CRG
previously discussed for recorder participants.
The CTI Event message is routed through the Scheduler, and is
altered by the Scheduler to indicate which participants re recorder
participants as well as which ones are selective record
participants. Recorder participants trigger the CRG to sweep any
audio from VOX MCRs that overlap in time. When the CRG detects an
overlap between recorder participant and selective record
participant times, the audio that is swept into the CTI MCR for
this overlap period is discarded. This causes the audio to be
removed from both VOX and CTI MCRs, which prevents any chance of
the audio being made available for playback or archive.
Selective Record Event: The Selective Record command is an event
originating from the Scheduler. It identifies either a participant
that is not to be recorded or that an entire call record should not
be recorded. In one embodiment the system is capable of handling
recording exceptions based upon information obtained from the CTI
data. Criteria for selective record processing are discussed
below.
Selective Record feature can take on two meanings. In one instance,
a customer may want to record all telephony events except for ones
that meet specific criteria. In a second instance, a customer may
only want to record calls that meet certain criteria.
Since selective recording can possibly be triggered from multiple
sources, in a preferred embodiment this decision process is located
in the Master Controller, a sub-component of the subject system's
Scheduling & Control Services.
Suggested reasons for not recording all or parts of a call are
based upon the following examples of CTI event data.
Event Data Explanation Results Agent exclusion based Supervisor
involved Delete audio for agent's upon participants calls not to be
included participation during the AgentID call, and the associated
references in the MCR. Exclusion based upon CEO involved calls
Delete audio for agent's Extension or fully not to be recorded.
participation during the qualified phone (whether at office or at
call, and the associated Number of participant. home) references in
the MCR. Combination of Prisoner calls his Delete all audio as well
AgentID of one par- lawyer. as the entire call record. ticipant and
fully qual- ified phone number of another participant
Based upon these conditions and any future rules established inside
the Master Controller (MC), exclusion can take place on audio
recorded during a target participant's time of involvement or over
the entire call record.
The chain of events involved in Selective Record (Call Exclusion)
is as follows:
1. Recorder detects presence of audio and records to audio
buffer.
2. Recorder sends VOX events to CRG indicating presence of
audio.
3. CRG creates new call record based upon VOX event.
4. The CTI server sends call events to CRG and MC.
5. CRG associates CTI event data with VOX based call record.
6. MC checks for selective record triggers based upon criteria
indicated above. If a criterion is met, a Selective Record
(exclusion) command is sent to both Recorder and CRG indicating the
start of the selective record interval.
7. Recorder deletes audio indicated in selective record message and
continues to suppress recording until instructed otherwise.
8. CRG alters the call record to eliminate details of participant
or deletes the call record.
9. Upon completion of the call, the CTI Server sends call events to
the CRG and MC.
10. MC checks for selective record triggers based upon criteria
indicated above. If a criterion is met, a selective record
(exclusion) is sent to the Recorder indicating the end of the
selective record interval.
11. The Recorder resumes its normal mode of audio recording.
Selective Record (Call Inclusion)
1. The CTI server sends call events to CRG and MC. CRG creates MCR
and populates with events. Since default is set not to record, the
flag m_bDontArchive is set to prevent the local data store from
writing it to the database.
2. MC checks for selective record triggers based upon criteria
indicated above. If a criterion is met, a Selective Record
(inclusion) command is sent to both Recorder and CRG indicating the
start of the selective record interval. CRG sets m_bDontArchive to
false and immediately instructs local data store to archive.
3. Recorder detects presence of audio and records to audio
buffer.
4. Recorder sends history of VOX events to CRG in a VoxSummary
message.
5. CRG creates new call record based upon VOX event.
6. CRG associates CTI event data with VOX based call record.
7. Upon completion of the call, the CTI Server sends call events to
the CRG and MC.
8. MC checks for selective record triggers based upon criteria
indicated above. If a criterion is met, a selective record
(inclusion) command is sent to the Recorder indicating the end of
the selective record interval.
9. The Recorder resumes its normal mode of suppressing the audio
recording.
The format of the Recorder's Selective Record command is shown
below.
Name Type Description StartTime time_and_date Start Time of
recording interval EndTime Time_and_date End Time of recording
interval bRecordAudio bool If true, record audio during the
indicated interval. If false, suppress any audio recording during
the indicated interval.
Since the recorder has no knowledge of participants or call record
boundaries, the MC needs to inform the recorder when to start a
selective record interval and when to stop. The boolean
bRecordAudio signifies what action should be taken during this
interval.
When an event occurs that triggers the start of a selective record
interval, the Recorder's selective record command informs the
recorder of the interval start. The End time is most likely not
known at this point so it is set to some invalid value in order to
indicate that audio should be recorded (or suppressed) for an
indefinite period until a subsequent command is received.
When an event occurs that triggers the end of a selective record
interval, the Recorder's selective record command informs the
recorder of the interval end. The End time indicates when the
selective record interval is complete. The recorder returns to its
normal recording mode based upon its original configuration.
Any selected audio committed to file needs to be removed from the
file and replaced with a silence entry for that period.
The format of the CRG selective record command is shown below.
Name Type Description MCR Number UUID MCR affected by this
selective record command Participant Index UINT Index of
participant in MCR not to be recorded. (if Reason=1) Reason BYTE 1
=Participant, 2=Entire Call
For the CRG, only a single event that indicates what is selectively
recorded is needed. If the Reason code indicates that the entire
call record is to be deleted, the CRG will mark the call record
such that it is removed from the database if it has already been
written or not logged in the first place. If selective record
affects a specific participant, the call record can either be left
unmodified (since the recorder has already handled deletion of
audio) or the participant can be overwritten to remove his/her
details.
The system configuration can be adjusted so that the CRG will
operate in either fashion, depending on whether removing the audio
alone is sufficient for the desired application of the system, or
if the metadata must also be removed to eliminate the records of
telephone numbers dialed, etc.
CRG Software Implementation
In the preferred embodiment of the subject system, the CRG is
implemented as an in-process COM DLL that is associated with the
Audio Recorder process, and therefore these two components reside
together upon the Voice Server. COM, here, is Common Object Model,
a distributed computing architecture designed by Microsoft
Corporation to facilitate cooperative processing among software
elements on a LAN. DLL is Dynamic Link Library, a means whereby
executable code can be encapsulated in a package that can be loaded
upon demand and shared by several programs, rather than being
packaged as a separate, isolated executable program. The Audio
Recorder process is responsible for creating the CRG COM object as
well as starting and stopping the CRG subsystem. The Data Store
module that interfaces with the CRG is a statically linked DLL.
Class Design
FIG. 19 illustrates the class diagram of the Call Record Generator.
The CRG module is itself comprised of a plurality of modules, as
shown in the figure, and explained below.
CallRecordEvent Processor--the CallRecordEventProcessor class 1912
is the main class of the CRG. It is instantiated during the
Initialize method call of the CRG interface. It is responsible for
allocating the rest of the CRG objects. On instantiation, it
acquires the channel count for the recorder (currently limited to
128) and instantiates a group a classes for each recording input
channel. These classes include a CallRecordChannelManager 1916 and
RecorderLocation 1920 for each channel. The
CallRecordEventProcessor 1912 creates the Recorder 1924 and Facade
1928 Event input queues. Reading and processing of configuration
information from the subject system's database takes place in the
CallRecordEventProcessor 1912. Events received that cause a change
in configuration are processed there.
CallRecordChannelManager--This class manages the call records for a
specific recording input channel. It is responsible for creating,
populating, and closing call records with event information
received from the CRG event processor. If event information is
deemed as significant, the CallRecordChannelManager 1916 will send
an event to the DataStoreEventQueue 1932 in order for the update to
be reflected in the local data store.
MasterCallRecord--This class 1936 holds information that is global
to an entire call. Global information includes identifiers for the
call record, the start and stop times of the entire call, the
recorder location with respect to the switch, and flags indicating
the call record status. It also contains a list of the participants
within a call, based upon information supplied by CTI events. It
acts as a centralized point of control for merging call record
information for a given telephone call.
VoxCallRecord--This class 1940 is a superclass of the
MasterCallRecord class 1936. It contains information dealing with
events provided by the recorder. It holds the details of a call,
such as the start/stop times, media filenames and other data that
can be supplied by the recorder.
RecorderLocation--This class 1920 holds the information relating a
logical device on a telephony switch with a specific Voice Server
and recording input channel.
The following table indicates configuration information needed by
the CRG at runtime.
Acceptable Configuration Field Type Values Default Description
sSysTimeCoupling String "TIGHT", "LOOSE" Indicates how time-based
recorder "LOOSE" and CTI events are compared to determine a match.
TIGHT - Recorder times must fit entirely inside CTI times for a
positive result. LOOSE - Recorder times need to overlap with CTI
times for a positive result. nCompleteCallHoldOver DWORD
0..42949672 90 - For Maximum number of seconds that Period 96
realtime a completed call record is kept in (in sec) CTI. (Much the
history list. This holdover larger if non- allows events coming
from realtime different sources to affect the call CTI) record
before it is made persistent. After this holdover period expires,
no more events can update the call record.
nActiveCallHoldoverPeriod DWORD 0..42949672 86400 The maximum
number of seconds a 96 (24 hours) call is allowed to exist before
being (in sec) forcibly closed. This is used as a safeguard against
missing CTI or Recorder events that would normally end a call
record. nMCRMaxSize WORD 0..65535 100 Maximum number of entries (in
entries) allowed in the MasterCall Record history list nSystemSkew
WORD 0..65535 0 A known, fixed difference (in (in sec) seconds)
that specifies the skew between a Recorder clock and a PBX clock.
Used to adjust incoming CTI event times before processing and
comparing with Recorder event times. ynCTIDataFromRecorder bool 1 =
yes yes Identifies, in cases where Recorder 0 = no and CTI
information overlaps, which source is preferred to populate the
call records. nSaveVoxClipsLonger WORD 0..65535 6 This setting is
used to avoid ThanSeconds creating VOX based call records from
noise on recording input channels. It directs the CRG to discard
any VOX clips that do not exceed the specified number of seconds in
duration.
Stream Control Manager
As noted above, in a preferred embodiment, the system of the
present invention taps into activity on a PBX (Private Branch
Exchange) by intercepting audio on either the trunk or extension
side of a phone call. The tapped audio is then redirected as input
to a channel on a DSP (Digital Signal Processor) based voice
processing board, which in turn is digitized and stored into
program-addressable buffers. The recorded audio is then combined
with descriptive information ("metadata") obtained through a
Computer Telephony Integration (CTI) communications link with the
PBX and stored as a single manageable unit ("Voicedata") to
facilitate its subsequent search and retrieval.
The preferred embodiment leverages Computer Telephony Integration,
to supplement the recorded audio data. As discussed above, CTI is
provided through a data link from specific telephone switching
equipment located at the customer site, which is then input to the
recording system's CTI Server. Supplied data includes such items as
telephone numbers of involved parties, caller ID/ANI information,
DNIS information, and agent ID numbers. The CTI Server performs the
task of analyzing and reorganizing data from both the real-time and
SMDR (asynchronous) links, and passing the results onwards to the
remainder of the recording system for further processing.
A module called the "Call Record Generator," or CRG, discussed
above, is then responsible for collecting data from the CTI Server,
creating `master call records` and attempting to match those
records with existing recorded audio data. If the CRG receives CTI
information indicating that audio data recorded on two Voice
Servers is related (for example, due to a transferred call),
records will be generated for each portion with a common call
record ID. This ID can later be used to query for all the pieces
(or "segments") comprising the complete call. In addition, each
segment will indicate the Voice Server which contains that piece of
the call.
During playback, the User Workstation's player module connects to a
program located on a Voice Server called the Playback Server, or
PBServer. The machine name of the particular Voice Server with
which a communications session should be established, stored by the
CRG in the call record table of the Voicedata storage module, is
passed into the player module after being extracted by the User
Workstation's call record browser. A call record playback request
is then submitted, which causes the PBServer to query for a
specific call record's audio files located on that physical
machine, open them, and prepare to stream the audio upon buffer
requests back to the client. If successful, a series of requests is
then issued from the client, each of which will obtain just enough
audio to play to a waveOut device while maintaining a safety net of
extra audio in case of network delays. Upon a request to "Move"
within the scope of a call record, the PBServer will reposition its
read pointer to the desired location and then begin passing back
buffers from that point. This series of Request and Move commands
will continue until the user chooses to end the session by shutting
down the client-side audio player.
When a call is transferred between locations, it is possible that
the call may span multiple Voice Servers, since the extensions or
trunks involved may be monitored by different recorders. If this is
the case, the audio data is spread out between playback servers,
and it must be properly pieced back together to reconstruct the
complete call for a playback client.
There are several possible solutions to the problem. First of all,
one could choose one central server and copy in all data from the
involved servers. This is as slow as copying the files locally to
the client, but it at least consolidates the data to one location
for the playback server to operate on. Assuming that this method is
chosen, however, several new problems arise. First is the issue of
drive space: depending on the number of transfers and recorders
involved with a call record, the central playback server could end
up suddenly storing a large number of files. This is multiplied by
the total number of clients requesting playback sessions. Soon
enough, a large amount of unpredictable space is being allocated
and freed without any reasonable way of estimating the space
necessary to service all requests. Similarly, the processor and
memory load on this server is taking the brunt of being used for
every playback request, since even normal, single recorder playback
sessions would be routed through this one machine.
Another solution would be to have the central playback server run
some intermediate process that would stream all of the data from
the multiple servers back to each client, like a "funnel." This
would avoid the copying and drive space issues, but there are still
two problems. First, the centralizing of this server once again
puts the entire load on a single machine. But more importantly, if
multiple streams are being funneled through this one location, the
server would somehow need to organize the streams so that during
playback, they appear to be arranged in the proper order.
The Stream Control Manager (SCM) used in accordance with a
preferred embodiment is the result of addressing the issues
referred to in the second solution discussed above. With regard to
the resource issue, the solution was to simply move the "funneling"
module from one central server to the client side. In this way,
servers are still providing the actual requested data, but it
becomes the client side's responsibility to bring the data
together. Yet the SCM remains a separate, COM-based module so
encapsulation is still maintained (a client application is not
hard-wired directly into the SCM code). This was intentional since
other system modules in alternate embodiments of the system need to
reuse the SCM to gather playback data (e.g., for phone handset
playback support instead of LAN playback support) or to gather
audio from a multitude of Voice Servers for long-term offline
storage on DAT or DVD media.
The process of stream management begins when the SCM is sent a list
of segments which comprise the entire call. Each segment includes
the machine name of the Voice Server, the segment's start time,
duration, channel ID, and an event callback routine provided by the
client which serves as a destination for the final organized
data.
Once this list is received and stored as a vector (array), the SCM
proceeds to try connecting to all servers required to play back
this call. The connection, if successfully established, is
associated with its respective segment via a pointer in the segment
entry. The connection is also added to an array so that if a
subsequent segment's server is the same as an earlier segment, the
connection can be reused. This may occur if a call transfers away
to a line monitored by a second recorder and is later transferred
back again to the original line. If the process cannot complete
successfully (i.e., if a Voice Server is malfunctioning), playback
is aborted to avoid skipping over any necessary data.
Next, the SCM goes through its list of segments and for each,
handshakes with its server through a series of function calls.
During this phase, the SCM informs each playback server of the
desired segment to stream back by providing its start time,
duration, channel ID using the parameter data that was passed in
earlier. Once again, if any part of the procedure fails, the entire
initialization (and thus playback) is aborted. At the completion of
this phase, every server should have loaded all the audio files
associated with their portion of the entire data stream. Each is
now ready for audio buffer requests.
The SCM then waits for a client to execute a "StartStream" call. In
a graphical interface, this would occur, for example, when a user
hits a Play button or begins a Save operation. Once this function
is called, a separate thread spawns which will handle the entire
process.
First, the current play position is checked to see which segment to
begin playing on (a Move operation, explained below, controls the
manual repositioning of this value). This is determined by looping
through all of the segments, adding each segment's duration to a
running total. When the current segment's duration added to the
total exceeds the play position, that is the segment which contains
the current play position.
Once this calculation is complete, a loop begins which starts from
the previously determined segment and proceeds through the rest of
the segment vector. For each segment, requests are formed for a
predetermined buffer size and sent to the associated server. Once a
buffer is returned, based on a flag configurable from the client,
the SCM will either directly send back this data or "slice" it for
the client first before returning it. Here, slicing refers to a
process of dividing the buffer into smaller buffers by a least
common multiple known as a block align; this is sometimes useful to
a client with a graphical component because the interface may need
to reflect the amount played in smaller subdivisions.
When it is detected that all data from a segment has been
requested, the SCM automatically steps to the next segment
(possibly located on a different Voice Server) and begins
requesting data from it instead. Because all Voice Servers are
pre-loaded with the data and "ready to go," this process takes
place in a fraction of a second, and the client does not sense any
gap in the audio data being returned. In fact, the only true method
for discerning the segment boundaries involves listening for
normal, audible indicators of a transfer being made (clicking,
ringing, or hearing the voice of a new participant) as provided
through the telephone switch environment.
At the close of a play session (e.g., the user hits Stop or Pause
in a typical audio playback GUI displayed in conjunction with the
GUI described in FIG. 16) a StopStream call is made to the SCM. The
thread in turn detects that the stopped state has been entered,
exits from the request loop code, and frees up any used resources.
Finally, it informs the client that a Stop event has occurred. If
the entire call record is played without calling StopStream, the
SCM performs the same exit and cleanup code, but informs the client
that a Done event has occurred instead.
Movement within the overall stream is straightforward, given the
aforementioned method that the SCM uses to determine which segment
to begin playing from. A global variable holds the total number of
milliseconds of audio data requested thus far. When a Move is
performed, the server containing the data at the destination
position is told to re-position itself, and the current play
position is reset. Now, once StartStream executes again, it will
initially start requesting from the server that was just moved to.
And because that server had also moved its position pointer ahead,
data will not be streamed from the beginning of the segment, but
from where the Move position fell within that segment. Thus
movement is a synchronized action completely transparent to the
client, who is, ultimately, only interested in treating the data as
a single stream. SCM Pseudo-code 1. Initialize receives segment
description data (start time, duration, etc.)
a.) Form a vector of all segments.
b.) Try to connect to all segments' servers.
c.) If there is an error connecting to any server, exit.
d.) Try to initialize each connected server.
e.) If there is an error initializing any server, exit. 2. If
StartStream received:
a.) Go through segment list. Find segment of current play
position.
b.) Starting with that segment, contact the associated server and
begin requesting buffers.
c.) If option set, divide up buffer into smaller chunks.
d.) Send buffer(s) to client via event callback.
e.) Repeat until all data requested for this segment on that
server.
f.) Repeat from step b. with next segment in list. 3. If Stop
received:
a.) Exit from request loop.
b.) Clean up used resources.
c.) Send "Stop" event back to client. 4. If Stop not received, but
all data from all segments played:
a.) Exit from request loop.
b.) Clean up used resources.
c.) Send "Done" event back to client. 5. Move received:
a.) Go through segment list. Find segment of desired play
position.
b.) Contact the associated server and reposition to that desired
position.
c.) Reset current play position variable to reflect change.
Detailed flow diagrams describing SCM operation are provided in
FIGS. 20, 20A, 20B, 21, 22, 22A, 22B, and 22C.
FIG. 20 illustrates the initialization process of the Stream
Control Manager. The Initialization Sequence begins when a user
enters the User Workstation playback software and at step 2010
queries for a recorded call record by desired criteria. At step
2012 a call record browser displays resulting call records. At step
2014 the user selects the desired record for playback. At step 2016
the browser invokes a PbkControlWin object: a dialog containing the
`player` ActiveX control.
At step 2020 the browser sends information to PbkControlWin about
all segments 5 comprising the call record. If at step 2024
immediate playback is not required, at step 2028 the entry is added
to a playlist for future playback, and at step 2030 SUCCESS is
returned. If at step 2024 immediate playback is required, at step
2032 the call record ID and segment list are forwarded to a GUI
Player module. At step 2038 (see FIG. 20A) the player module
instantiates a local SCM (StreamControl) object and stores a
pointer in m_pIStreamControl. At step 2040 the player module
accepts the data, displays starting time and total duration (by
parsing out string data), and forwards it to the final module, the
Stream Control Manager (SCM), for audio playback.
Step 2046 begins the creation of a segments vector. At step 2046, a
segment is parsed out from segList. At step 2048, recorder ID,
start time, duration, and channel are parsed out from the segment.
At step 2050, a new SEGMENT structure is created from recorder ID,
start time, duration, and channel. At step 2052, a new SEGMENT is
added to the SEGMENT vector. At step 2054, if all segments have
been parsed from segList, at step 2058 an element is gotten from
the SEGMENT vector. If at step 2054 more segments remain to be
parsed from segList, steps 2046, 2048, 2050, and 2052 are
repeated.
After step 2058, the program determines at step 2060 whether a new
DCOM connection is required to the recorder for this segment. If
not, at step 2062 the existing pointer is copied from the
Connections vector to the server pointer in the SEGMENT vector and
the program proceeds to step 2076. If at step 2060 the connection
is new, a connection is made to the indicated recorder's
"PlayBackServer" DCOM object using CoCreateInstanceEx. At step 2066
the program checks whether the object instantiated successfully. If
not, at step 2068 a log error message occurs and at step 2070 ERROR
(C) is returned. If at step 2066 the object instantiated
successfully, at step 2072 (see FIG. 20B) the new object's pointer
is added to the Connections vector. At step 2074 the program
determines whether all segments have been connected. If not, the
program returns to step 2058. If at step 2074 all segments have
been connected, at step 2076 an element is gotten from the SEGMENT
vector. At step 2078 the program queries for a list of wave files
on the server that go with this segment. At step 2080 the program
determines whether the query was successful. If not, at step 2082 a
log error message occurs, and at step 2084 ERROR (C) is
returned.
If at step 2080 the query was successful, at step 2088 the program
opens the wave files on the server and prepares them for streaming.
It also returns the wave format of the audio in the segment. At
step 2093 the program determines whether the wave files and format
were obtained successfully. If not, at step 2094 a log error
message occurs and at step 2095 ERROR (C) is returned. If step 2088
is determined at step 2093 to have been successful, at step 2096
the program checks whether all segments have been initialized. If
not, the program returns to step 2076. If so, step 2097 is
performed and at step 2098 SUCCESS is returned.
FIG. 21 illustrates how the program manages a Player Object 2110
and a PbkControlWin Object 2132.
FIG. 22 illustrates the playback sequence of the Stream Control
Manager. Initially, at step 2202 a user has completed
initialization and is waiting to hit Play in the Player GUI. At
step 2204 the user hits the Play button. At step 2206 a message is
sent to the Play method in the Player ActiveX control. At step 2210
the Play method in Player ActiveX control causes the output buffers
to be "sliced" to increase the number of smaller buffers sent, thus
increasing the resolution of the "totalPlayed" variable. At step
2218 Play method causes the server-side position to move to the
current slider position. At step 2222 the program gets segment i++
from the SEGMENT vector. At step 2224 (see FIG. 22A) the program
determines whether the End Time offset for segment i is greater
than curPosition. If not, the program returns to step 2222. If so,
the program proceeds to step 2226 and causes the file pointer on
the server side to change to the appropriate new location. The
program checks at step 2230 whether step 2226 was successful. If
not, at step 2232 a log error message occurs and at step 2234 ERROR
(C) is returned.
If at step 2230 step 2226 is determined to have been successful, at
step 2238 the program calls Stream Control::StartStream. At step
2242 the program gets segment i++ from the SEGMENT vector. At step
2244 the program calls CoMarshalInterThreadInterfaceInStream to
marshal a DCOM pointer member across a thread boundary. At step
2246 the program determines whether all SEGMENT elements have been
marshaled. If not, the program returns to step 2242. If so, at step
2248 the main SCM streaming thread is spawned.
FIG. 22B illustrates an SCM main streaming thread. When the thread
begins, at step 2250 the thread gets a segment from the SEGMENT
vector. At step 2252 CoCetInterfaceAndReleaseStream is called to
unmarshal a DCOM pointer member across the thread boundary. At step
2254 the thread checks whether all SEGMENT elements have been
unmarshaled. If not, the thread returns to step 2250. If at step
2250 all SEGMENT elements are determined to have been unmarshaled,
at step 2256 the thread gets a segment from the SEGMENT vector. The
thread then checks at step 2258 whether the End Time offset for
segment i is greater than curAmountRequested. If not, the thread
returns to step 2256. If so, at step 2260 the thread gets
Segment[i++]. The thread checks at step 2262 whether i is less than
the highest segment number. If not, an Event::Done method is called
at step 2264, and at step 2266 SUCCESS (C) is returned. If so, at
step 2268 the thread determines whether this is the first segment
to be played in this instance of the thread. If not, at step 2270
the thread calls PBServer::PositionPLay(totalRequested) for
Segment[i] and goes to step 2272. If so, the thread goes directly
to step 2272.
At step 2272, the thread checks whether totalRequested is less than
Segment[i].endTimeOffset. If not, the thread returns to step 2260.
If so, the thread proceeds to step 2274 and checks whether
totalRequested plus bufferSize is less than or equal to
Segment[i].endTimeOffset. If not, at step 2276 the thread
calculates a new bufferSize in multiples of the audio format's
"block align." and proceeds to step 2278 (see FIG. 22C). If so, the
thread proceeds directly to step 2278. At step 2278, the thread
calls PBServer::ReqBuffer for Segment[i]. This is the core routine
that actually retrieves a buffer of data from the PlayBack Server.
At step 2286 the thread checks whether step 2278 was successful. If
not, at step 2284 a log error message occurs, and at step 2282
ERROR (C) is returned.
If at step 2286 the thread determines that step 2278 was
successful, at step 2287 toatIRequested is set equal to
totalRequested plus Actual returned buffer size. At step 2288, the
thread checks whether Blockslicing is enabled. If not, at step 2289
the thread sends the buffer back to the Player via Event::SendData
method and returns to step 2274. If BlockSlicing has been enabled,
at step 2292 the thread checks whether the CODEC is Dialogic OKI
ADPCM or PCM. If not, at step 2293 the slice of the slices is set
equal to the audio format's block align and the thread proceeds to
step 2296. If so, at step 2294 the size of the slices is set to an
even dividend of the buffer size (e.g., one-tenth of the buffer
size). At step 2296, the thread copies out "slice size" from the
buffer and sends it back to Player via Event;;SendData method. At
step 2298 the thread checks whether the entire buffer has been sent
back. If not, the thread returns to step 2298. If so, the thread
returns to step 2274.
The Stream Control Manager could theoretically be adapted to be
used in more general streaming media situations, outside that of
communications recording systems. In most current stream-based
systems for network-based playback of audio content, such as
RealMedia and NetShow, two general broadcast architectures exist
known as unicast and multicast. Unicast involves a single
client-server connection for data streaming, while in the multicast
scenario a server pushes data to a single network address which
multiple clients can then "tune in" to. However both models assume
that data is being continuously fed from a single server. In the
interest of load balancing, or if pieces of a streaming
presentation were spread out across multiple locations, the SCM
model could provide an innovative solution where the client side
has the power to weave together many streams into a single playback
session. An example could be imagined where a news organization,
such as CNN, dynamically assembles a streaming broadcast for the
online viewer from many different reports located on servers across
the country. The components could be played seamlessly end-on-end
using the SCM model, and if the viewer desired to rewind or
fast-forward to a specific point in the stream, the SCM model would
allow for complete transparent control.
The present invention is not to be limited in scope by the specific
embodiments described herein. Indeed, modifications of the
preferred embodiment in addition to those described herein will
become apparent to those skilled in the art from the foregoing
description and accompanying figures. Doubtless, numerous other
embodiments can be conceived that would not depart from the
teaching of the present invention, which scope is defined by the
following claims.
All the features disclosed in this specification (including any
accompanying claims, abstract, and drawings) may be replaced by
alternative features serving the same, equivalent, or similar
purpose, unless expressly stated otherwise. Thus, unless expressly
stated otherwise, each feature disclosed is one example only of a
generic series of equivalent or similar features.
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